Prediction of absolute thresholds and equal-loudness contours using a modified loudness model

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.

Causal inference in environmental sound recognition

Sound is caused by physical events in the world. Do humans infer these causes when recognizing sound sources? We tested whether the recognition of common environmental sounds depends on the inference of a basic physical variable - the source intensity (i.e., the power that produces a sound). A source's intensity can be inferred from the intensity it produces at the ear and its distance, which is normally conveyed by reverberation. Listeners could thus use intensity at the ear and reverberation to constrain recognition by inferring the underlying source intensity. Alternatively, listeners might separate these acoustic cues from their representation of a sound's identity in the interest of invariant recognition. We compared these two hypotheses by measuring recognition accuracy for sounds with typically low or high source intensity (e.g., pepper grinders vs. trucks) that were presented across a range of intensities at the ear or with reverberation cues to distance. The recognition of low-intensity sources (e.g., pepper grinders) was impaired by high presentation intensities or reverberation that conveyed distance, either of which imply high source intensity. Neither effect occurred for high-intensity sources. The results suggest that listeners implicitly use the intensity at the ear along with distance cues to infer a source's power and constrain its identity. The recognition of real-world sounds thus appears to depend upon the inference of their physical generative parameters, even generative parameters whose cues might otherwise be separated from the representation of a sound's identity.

Convergent and Distinct Effects of Multisensory Combination on Statistical Learning Using a Computer Glove

Learning to play a musical instrument involves mapping visual + auditory cues to motor movements and anticipating transitions. Inspired by the serial reaction time task and artificial grammar learning, we investigated explicit and implicit knowledge of statistical learning in a sensorimotor task. Using a between-subjects design with four groups, one group of participants were provided with visual cues and followed along by tapping the corresponding fingertip to their thumb, while using a computer glove. Another group additionally received accompanying auditory tones; the final two groups received sensory (visual or visual + auditory) cues but did not provide a motor response—all together following a 2 × 2 design. Implicit knowledge was measured by response time, whereas explicit knowledge was assessed using probe tests. Findings indicate that explicit knowledge was best with only the single modality, but implicit knowledge was best when all three modalities were involved.

No Effect of Musical Training on Frequency Selectivity Estimated Using Three Methods

It is widely believed that the frequency selectivity of the auditory system is largely determined by processes occurring in the cochlea. If so, musical training would not be expected to influence frequency selectivity. Consistent with this, auditory filter shapes for low center frequencies do not differ for musicians and nonmusicians. However, it has been reported that psychophysical tuning curves (PTCs) at 4000 Hz were sharper for musicians than for nonmusicians. This study explored the origin of the discrepancy across studies. Frequency selectivity was estimated for musicians and nonmusicians using three methods: fast PTCs with a masker that swept in frequency, "traditional" PTCs obtained using several fixed masker center frequencies, and the notched-noise method. The signal frequency was 4000 Hz. The data were fitted assuming that each side of the auditory filter had the shape of a rounded-exponential function. The sharpness of the auditory filters, estimated as the Q₁₀ values, did not differ significantly between musicians and nonmusicians for any of the methods, but detection efficiency tended to be higher for the musicians. This is consistent with the idea that musicianship influences auditory proficiency but does not influence the peripheral processes that determine the frequency selectivity of the auditory system.

Temporal weights in loudness: Investigation of the effects of background noise and sound level

Previous research has consistently shown that for sounds varying in intensity over time, the beginning of the sound is of higher importance for the perception of loudness than later parts (primacy effect). However, in all previous studies, the target sounds were presented in quiet, and at a fixed average sound level. In the present study, temporal loudness weights for a time-varying narrowband noise were investigated in the presence of a continuous bandpass-filtered background noise and the average sound levels of the target stimuli were varied across a range of 60 dB. Pronounced primacy effects were observed in all conditions and there were no significant differences between the temporal weights observed in the conditions in quiet and in background noise. Within the conditions in background noise, there was a significant effect of the sound level on the pattern of weights, which was mainly caused by a slight trend for increased weights at the end of the sounds ("recency effect") in the condition with lower average level. No such effect was observed for the in-quiet conditions. Taken together, the observed primacy effect is largely independent of masking as well as of sound level. Compatible with this conclusion, the observed primacy effects in quiet and in background noise can be well described by an exponential decay function using parameters based on previous studies. Simulations using a model for the partial loudness of time-varying sounds in background noise showed that the model does not predict the observed temporal loudness weights.

Physiologically motivated individual loudness model for normal hearing and hearing impaired listeners

A loudness model with a central gain is suggested to improve individualized predictions of loudness scaling data from normal hearing and hearing impaired listeners. The current approach is based on the loudness model of Pieper et al. [(2016). J. Acoust. Soc. Am. 139, 2896], which simulated the nonlinear inner ear mechanics as transmission-line model in a physical and physiological plausible way. Individual hearing thresholds were simulated by a cochlear gain reduction in the transmission-line model and linear attenuation (damage of inner hair cells) prior to an internal threshold. This and similar approaches of current loudness models that characterize the individual hearing loss were shown to be insufficient to account for individual loudness perception, in particular at high stimulus levels close to the uncomfortable level. An additional parameter, termed "post gain," was introduced to improve upon the previous models. The post gain parameter amplifies the signal parts above the internal threshold and can better account for individual variations in the overall steepness of loudness functions and for variations in the uncomfortable level which are independent of the hearing loss. The post gain can be interpreted as a central gain occurring at higher stages as a result of peripheral deafferentation.

Development of a model to predict the likelihood of complaints due to assorted tone-in-noise combinations

This paper develops a model to predict if listeners would be likely to complain due to annoyance when exposed to a certain noise signal with a prominent tone, such as those commonly produced by heating, ventilation, and air-conditioning systems. Twenty participants completed digit span tasks while exposed in a controlled lab to noise signals with differing levels of tones, ranging from 125 to 1000 Hz, and overall loudness. After completing the digit span tasks under each noise signal, from which task accuracy and speed of completion were captured, subjects were asked to rate level of annoyance and indicate the likelihood of complaining about the noise. Results show that greater tonality in noise has statistically significant effects on task performance by increasing the time it takes for participants to complete the digit span task; no statistically significant effects were found on task accuracy. A logistic regression model was developed to relate the subjective annoyance responses to two noise metrics, the stationary Loudness and Tonal Audibility, selected for the model due to high correlations with annoyance responses. The percentage of complaints model showed better performance and reliability over the percentage of highly annoyed or annoyed.

A Loudness Model for Time-Varying Sounds Incorporating Binaural Inhibition

This article describes a model of loudness for time-varying sounds that incorporates the concept of binaural inhibition, namely, that the signal applied to one ear can reduce the internal response to a signal at the other ear. For each ear, the model includes the following: a filter to allow for the effects of transfer of sound through the outer and middle ear; a short-term spectral analysis with greater frequency resolution at low than at high frequencies; calculation of an excitation pattern, representing the magnitudes of the outputs of the auditory filters as a function of center frequency; application of a compressive nonlinearity to the output of each auditory filter; and smoothing over time of the resulting instantaneous specific loudness pattern using an averaging process resembling an automatic gain control. The resulting short-term specific loudness patterns are used to calculate broadly tuned binaural inhibition functions, the amount of inhibition depending on the relative short-term specific loudness at the two ears. The inhibited specific loudness patterns are summed across frequency to give an estimate of the short-term loudness for each ear. The overall short-term loudness is calculated as the sum of the short-term loudness values for the two ears. The long-term loudness for each ear is calculated by smoothing the short-term loudness for that ear, again by a process resembling automatic gain control, and the overall loudness impression is obtained by summing the long-term loudness across ears. The predictions of the model are more accurate than those of an earlier model that did not incorporate binaural inhibition.

Testing and refining a loudness model for time-varying sounds incorporating binaural inhibition

This paper describes some experimental tests and modifications to a model of loudness for time-varying sounds incorporating the concept of binaural inhibition. Experiment 1 examined the loudness of a 100% sinusoidally amplitude-modulated 1000-Hz sinusoidal carrier as a function of the interaural modulation phase difference (IMPD). The IMPD of the test sound was 90° or 180° and that of the comparison sound was 0°. The level difference between the test and the comparison sounds at the point of equal loudness (the LDEL) was estimated for baseline levels of 30 and 70 dB sound pressure level and modulation rates of 1, 2, 4, 8, 16, and 32 Hz. The LDELs were negative (mean = -1.1 and -1.5 dB for IMPDs of 90° and 180°), indicating that non-zero IMPDs led to increased loudness. The original version of the model predicted the general form of the results, but there were some systematic errors. Modifications to the time constants of the model gave a better fit to the data. Experiment 2 assessed the loudness of unintelligible speech-like signals, generated using a noise vocoder, whose spectra and time pattern differed at the two ears. Both the original and modified models gave good fits to the data.

Temporal weights in the perception of sound intensity: Effects of sound duration and number of temporal segments

Loudness is a fundamental aspect of auditory perception that is closely related to the physical level of the sound. However, it has been demonstrated that, in contrast to a sound level meter, human listeners do not weight all temporal segments of a sound equally. Instead, the beginning of a sound is more important for loudness estimation than later temporal portions. The present study investigates the mechanism underlying this primacy effect by varying the number of equal-duration temporal segments (5 and 20) and the total duration of the sound (1.0 to 10.0 s) in a factorial design. Pronounced primacy effects were observed for all 20-segment sounds. The temporal weights for the five-segment sounds are similar to those for the 20-segment sounds when the weights of the segments covering the same temporal range as a segment of the five-segment sounds are averaged. The primacy effect can be described by an exponential decay function with a time constant of about 200 ms. Thus, the temporal weight assigned to a specific temporal portion of a sound is determined by the time delay between sound onset and segment onset rather than by the number of segments or the total duration of the sound.

Extending sharpness calculation for an alternative loudness metric input

Sound quality metrics help improve the psychoacoustic acceptability of devices and environments by modeling and thus enabling deliberate improvement of perceptual attributes. Sharpness as defined in DIN 45692 [(2009). Deutsches Institut für Normung, Berlin] requires inputs from Zwicker's loudness metric [ISO 532-1 (2017). International Organization for Standardization, Geneva]. This letter demonstrates that sharpness can be formulated to accept specific loudness values from Moore and Glasberg's loudness metric [ISO 532-2 (2017). International Organization for Standardization, Geneva; ANSI S3.4 (2007). American National Standards Institute, Inc., Washington, DC]. Sharpness calculations using the two loudness metrics produce similar results. This method thus enables evaluation of sharpness as a straightforward add-on to standard loudness calculations using Moore and Glasberg's metric, for which sharpness calculations were not previously available.

The detailed shapes of equal-loudness-level contours at low frequencies

High-resolution equal-loudness-level contours (ELCs) were measured over the frequency range 10-250 Hz using 19 normal-hearing subjects. Three levels of the 50-Hz reference sound were used, corresponding to the levels at 50 Hz of the 30-, 50-, and 70-phon standardized ELCs given in ISO-226:2003. The dynamic range of the contours generally decreased with increasing reference level, and the slope was shallow between 10 and 20 Hz, consistent with previous studies. For the lowest level, the ELCs were sometimes but not always smooth and on average followed the standardized 30-phon contour for frequencies above 40 Hz. For the two higher levels, the individual ELCs showed a distinct non-monotonic feature in a "transition region" between about 40 and 100 Hz, where the slope could reach near-zero or even positive values. The pattern of the non-monotonic feature was similar across levels for the subjects for whom it was observed, but the pattern varied across subjects. Below 40 Hz, the slopes of the ELCs increased markedly for all loudness levels, and the levels exceeded those of the standardized ELCs. Systematic deviations from the standardized ELCs were largest for frequencies below 40 Hz for all levels and within the transition region for the two higher levels.

Detecting temporal changes in acoustic scenes: The variable benefit of selective attention

Four experiments investigated change detection in acoustic scenes consisting of a sum of five amplitude-modulated pure tones. As the tones were about 0.7 octave apart and were amplitude-modulated with different frequencies (in the range 2-32 Hz), they were perceived as separate streams. Listeners had to detect a change in the frequency (experiments 1 and 2) or the shape (experiments 3 and 4) of the modulation of one of the five tones, in the presence of an informative cue orienting selective attention either before the scene (pre-cue) or after it (post-cue). The changes left intensity unchanged and were not detectable in the spectral (tonotopic) domain. Performance was much better with pre-cues than with post-cues. Thus, change deafness was manifest in the absence of an appropriate focusing of attention when the change occurred, even though the streams and the changes to be detected were acoustically very simple (in contrast to the conditions used in previous demonstrations of change deafness). In one case, the results were consistent with a model based on the assumption that change detection was possible if and only if attention was endogenously focused on a single tone. However, it was also found that changes resulting in a steepening of amplitude rises were to some extent able to draw attention exogenously. Change detection was not markedly facilitated when the change produced a discontinuity in the modulation domain, contrary to what could be expected from the perspective of predictive coding.

Altered attentional filters in subjects with graded levels of sensorineural hearing loss

Near-threshold tones (targets) in noise that are preceded by cues of the same frequency or occur with a high probability are detected better than tones of other frequencies that may occur with a lower probability (probes); the better detection of targets than probes defines the attentional filter. We measured attentional filters using a cued probe-signal procedure with a two-interval forced-choice (2IFC) method in normal-hearing subjects (N = 15) and subjects with sensorineural hearing loss (SNHL; N = 14) with a range of hearing levels. Attentional filters were altered in SNHL subjects, who detected low-frequency probes as well as targets at all hearing levels and who detected high-frequency probes increasingly well with increasing hearing level. These effects were present in both intervals of the 2IFC procedure. As auditory filters measured psychophysically are typically asymmetric in subjects with SNHL, these results suggest that the signal frequencies affected by the attentional filter are governed by the shapes of the auditory filters at and around the cue frequency. The normal-hearing subjects showed the expected attentional filters in the first interval and shallower filters in the second interval, suggesting that the cue-evoked attentional process is transient. In the first interval, both low- and high-frequency probes were detected better as hearing level increased over a narrow range (from -5 to 10 dB at the target frequency), with a resultant loss of attentional filtering. This finding adds to observations of variable auditory function in individuals with clinically normal hearing thresholds established by audiometry.

A review of the history, development and application of auditory weighting functions in humans and marine mammals

This document reviews the history, development, and use of auditory weighting functions for noise impact assessment in humans and marine mammals. Advances from the modern era of electroacoustics, psychophysical studies of loudness, and other related hearing studies are reviewed with respect to the development and application of human auditory weighting functions, particularly A-weighting. The use of auditory weighting functions to assess the effects of environmental noise on humans-such as hearing damage-risk criteria-are presented, as well as lower-level effects such as annoyance and masking. The article also reviews marine mammal auditory weighting functions, the development of which has been fundamentally directed by the objective of predicting and preventing noise-induced hearing loss. Compared to the development of human auditory weighting functions, the development of marine mammal auditory weighting functions have faced additional challenges, including a large number of species that must be considered, a lack of audiometric information on most species, and small sample sizes for nearly all species for which auditory data are available. The review concludes with research recommendations to address data gaps and assumptions underlying marine mammal auditory weighting function design and application.

The effect of the helicotrema on low-frequency loudness perception

Below approximately 40 Hz, the cochlear travelling wave reaches the apex, and differential pressure is shunted through the helicotrema, reducing hearing sensitivity. Just above this corner frequency, a resonance feature is often observed in objectively measured middle-ear-transfer functions (METFs). This study inquires whether overall and fine structure characteristics of the METF are also perceptually evident. Equal-loudness-level contours (ELCs) were measured between 20 and 160 Hz for 14 subjects in a purpose-built test chamber. In addition, the inverse shapes of their METFs were obtained by adjusting the intensity of a low-frequency suppressor tone to maintain an equal suppression depth of otoacoustic emissions for various suppressor tone frequencies (20-250 Hz). For 11 subjects, the METFs showed a resonance. Six of them had coinciding features in both ears, and also in their ELC. For two subjects only the right-ear METF was obtainable, and in one case it was consistent with the ELC. One other subject showed a consistent lack of the feature in their ELC and in both METFs. Although three subjects displayed clear inconsistencies between both measures, the similarity between inverse METF and ELC for most subjects shows that the helicotrema has a marked impact on low-frequency sound perception.

Effectiveness of a loudness model for time-varying sounds in equating the loudness of sentences subjected to different forms of signal processing

A model for the loudness of time-varying sounds [Glasberg and Moore (2012). J. Audio. Eng. Soc. 50, 331-342] was assessed for its ability to predict the loudness of sentences that were processed to either decrease or increase their dynamic fluctuations. In a paired-comparison task, subjects compared the loudness of unprocessed and processed sentences that had been equalized in (1) root-mean square (RMS) level; (2) the peak long-term loudness predicted by the model; (3) the mean long-term loudness predicted by the model. Method 2 was most effective in equating the loudness of the original and processed sentences.

Physiological motivated transmission-lines as front end for loudness models

The perception of loudness is strongly influenced by peripheral auditory processing, which calls for a physiologically correct peripheral auditory processing stage when constructing advanced loudness models. Most loudness models, however, rather follow a functional approach: a parallel auditory filter bank combined with a compression stage, followed by spectral and temporal integration. Such classical loudness models do not allow to directly link physiological measurements like otoacoustic emissions to properties of their auditory filterbank. However, this can be achieved with physiologically motivated transmission-line models (TLMs) of the cochlea. Here two active and nonlinear TLMs were tested as the peripheral front end of a loudness model. The TLMs are followed by a simple generic back end which performs integration of basilar-membrane "excitation" across place and time to yield a loudness estimate. The proposed model approach reaches similar performance as other state-of-the-art loudness models regarding the prediction of loudness in sones, equal-loudness contours (including spectral fine structure), and loudness as a function of bandwidth. The suggested model provides a powerful tool to directly connect objective measures of basilar membrane compression, such as distortion product otoacoustic emissions, and loudness in future studies.

Human Pupillary Dilation Response to Deviant Auditory Stimuli: Effects of Stimulus Properties and Voluntary Attention

A unique sound that deviates from a repetitive background sound induces signature neural responses, such as mismatch negativity and novelty P3 response in electro-encephalography studies. Here we show that a deviant auditory stimulus induces a human pupillary dilation response (PDR) that is sensitive to the stimulus properties and irrespective whether attention is directed to the sounds or not. In an auditory oddball sequence, we used white noise and 2000-Hz tones as oddballs against repeated 1000-Hz tones. Participants' pupillary responses were recorded while they listened to the auditory oddball sequence. In Experiment 1, they were not involved in any task. Results show that pupils dilated to the noise oddballs for approximately 4 s, but no such PDR was found for the 2000-Hz tone oddballs. In Experiments 2, two types of visual oddballs were presented synchronously with the auditory oddballs. Participants discriminated the auditory or visual oddballs while trying to ignore stimuli from the other modality. The purpose of this manipulation was to direct attention to or away from the auditory sequence. In Experiment 3, the visual oddballs and the auditory oddballs were always presented asynchronously to prevent residuals of attention on to-be-ignored oddballs due to the concurrence with the attended oddballs. Results show that pupils dilated to both the noise and 2000-Hz tone oddballs in all conditions. Most importantly, PDRs to noise were larger than those to the 2000-Hz tone oddballs regardless of the attention condition in both experiments. The overall results suggest that the stimulus-dependent factor of the PDR appears to be independent of attention.

Effects of relative and absolute frequency in the spectral weighting of loudness

The loudness of broadband sound is often modeled as a linear sum of specific loudness across frequency bands. In contrast, recent studies using molecular psychophysical methods suggest that low and high frequency components contribute more to the overall loudness than mid frequencies. In a series of experiments, the contribution of individual components to the overall loudness of a tone complex was assessed using the molecular psychophysical method as well as a loudness matching task. The stimuli were two spectrally overlapping ten-tone complexes with two equivalent rectangular bandwidth spacing between the tones, making it possible to separate effects of relative and absolute frequency. The lowest frequency components of the "low-frequency" and the "high-frequency" complexes were 208 and 808 Hz, respectively. Perceptual-weights data showed emphasis on lowest and highest frequencies of both the complexes, suggesting spectral-edge related effects. Loudness matching data in the same listeners confirmed the greater contribution of low and high frequency components to the overall loudness of the ten-tone complexes. Masked detection thresholds of the individual components within the tone complex were not correlated with perceptual weights. The results show that perceptual weights provide reliable behavioral correlates of relative contributions of the individual frequency components to overall loudness of broadband sounds.

Subjective and objective rating of spectrally different pseudorandom noises--implications for speech masking design

Artificial sound masking is increasingly used in open-plan offices to improve speech privacy and to reduce distraction caused by speech sounds. Most of the masking sounds are based on pseudorandom continuous noise filtered to a specific spectrum that should be optimized in respect with speech masking efficiency and comfort. The aim of this study was to increase basic understanding regarding the comfort. The second aim was to determine how well objective rating methods (15 different noise indices) predict the subjective ratings. Twenty-three subjects rated the loudness, disturbance, pleasantness, and six other subjective measures of 11 spectrally different noises in laboratory conditions. Speech was not present during the experiment. All sounds were presented at 42 dB LAeq within 50-10,000 Hz. Unexpectedly, the subjects were most satisfied with sounds having emphasis on low frequencies. A sound having a slope of -7 dB per octave increment resulted in the highest satisfaction. Changes in subjective ratings were reasonably well predicted by five noise indices, while many well-known noise indices frequently used in building design underperformed in this task. The results are expected to benefit in the design of masking sounds and other appliances.

Development and current status of the "Cambridge" loudness models

This article reviews the evolution of a series of models of loudness developed in Cambridge, UK. The first model, applicable to stationary sounds, was based on modifications of the model developed by Zwicker, including the introduction of a filter to allow for the effects of transfer of sound through the outer and middle ear prior to the calculation of an excitation pattern, and changes in the way that the excitation pattern was calculated. Later, modifications were introduced to the assumed middle-ear transfer function and to the way that specific loudness was calculated from excitation level. These modifications led to a finite calculated loudness at absolute threshold, which made it possible to predict accurately the absolute thresholds of broadband and narrowband sounds, based on the assumption that the absolute threshold corresponds to a fixed small loudness. The model was also modified to give predictions of partial loudness—the loudness of one sound in the presence of another. This allowed predictions of masked thresholds based on the assumption that the masked threshold corresponds to a fixed small partial loudness. Versions of the model for time-varying sounds were developed, which allowed prediction of the masked threshold of any sound in a background of any other sound. More recent extensions incorporate binaural processing to account for the summation of loudness across ears. In parallel, versions of the model for predicting loudness for hearing-impaired ears have been developed and have been applied to the development of methods for fitting multichannel compression hearing aids.

Measurement and modeling of binaural loudness summation for hearing-impaired listeners

The summation of loudness across ears is often studied by measuring the level difference required for equal loudness (LDEL) of monaural and diotic sounds. Typically, the LDEL is ∼5-6 dB, consistent with the idea that a diotic sound is ∼1.5 times as loud as the same sound presented monaurally at the same level, as predicted by the loudness model of Moore and Glasberg [J. Acoust. Soc. Am. 121, 1604-1612 (2007)]. One might expect that the LDEL would be <5-6 dB for hearing-impaired listeners, because loudness recruitment leads to a more rapid change of loudness for a given change in level. However, previous data sometimes showed similar LDEL values for normal-hearing and hearing-impaired listeners. Here, the LDEL was measured for hearing-impaired listeners using narrowband and broadband noises centered at 500 Hz, where audiometric thresholds were near-normal, and at 3000 or 4000 Hz, where audiometric thresholds were elevated. The mean LDEL was 5.6 dB at 500 Hz and 4.2 dB at the higher center frequencies. The results were predicted reasonably well by an extension of the loudness model of Moore and Glasberg.

Effects of compression and onset/offset asynchronies on the detection of one tone in the presence of another

The weaker of two temporally overlapping complex tones can be easier to hear when the tones are asynchronous than when they are synchronous. This study explored how the use of fast and slow five-channel amplitude compression, as might be used in hearing aids, affected the ability to use onset and offset asynchronies to detect one (signal) complex tone when another (masking) complex tone was presented almost simultaneously. A 2:1 compression ratio was used with normal-hearing subjects, and individual compression ratios and gains recommended by the CAM2 hearing aid fitting method were used for hearing-impaired subjects. When the signal started before the masker, there was a benefit of compression for both normal-hearing and hearing-impaired subjects. When the signal finished after the masker, there was a benefit of fast compression for the normal-hearing subjects but no benefit for most of the hearing-impaired subjects, except when the offset asynchrony was relatively large (100 ms). The benefit of compression probably occurred because the compression improved the effective signal-to-masker ratio, hence reducing backward and forward masking. This apparently outweighed potential deleterious effects of distortions in envelope shape and the introduction of partially correlated envelopes of the signal and masker.

Cortical pitch regions in humans respond primarily to resolved harmonics and are located in specific tonotopic regions of anterior auditory cortex

Pitch is a defining perceptual property of many real-world sounds, including music and speech. Classically, theories of pitch perception have differentiated between temporal and spectral cues. These cues are rendered distinct by the frequency resolution of the ear, such that some frequencies produce "resolved" peaks of excitation in the cochlea, whereas others are "unresolved," providing a pitch cue only via their temporal fluctuations. Despite longstanding interest, the neural structures that process pitch, and their relationship to these cues, have remained controversial. Here, using fMRI in humans, we report the following: (1) consistent with previous reports, all subjects exhibited pitch-sensitive cortical regions that responded substantially more to harmonic tones than frequency-matched noise; (2) the response of these regions was mainly driven by spectrally resolved harmonics, although they also exhibited a weak but consistent response to unresolved harmonics relative to noise; (3) the response of pitch-sensitive regions to a parametric manipulation of resolvability tracked psychophysical discrimination thresholds for the same stimuli; and (4) pitch-sensitive regions were localized to specific tonotopic regions of anterior auditory cortex, extending from a low-frequency region of primary auditory cortex into a more anterior and less frequency-selective region of nonprimary auditory cortex. These results demonstrate that cortical pitch responses are located in a stereotyped region of anterior auditory cortex and are predominantly driven by resolved frequency components in a way that mirrors behavior.

Towards a method for loudness-based analysis of the sound of one's own voice

This paper outlines the steps in objectively estimating the time-varying loudness of one's own voice in a room (i.e. autophonic loudness). Voice recordings, made with a near-mouth microphone, are converted to the sound that reaches the two eardrums of the talking (or singing)-listener by convolving them with the impulse responses from the mouth to the respective ears of an anthropomorphic head and torso. The influences of bone-conducted sound and room reflections are taken into account. These convolved recordings are then processed with a computational time-varying loudness model. The method is demonstrated by a short case study, and the results illustrate something of the benefit of loudness analysis over sound pressure level analysis for representing autophonic loudness.

The perceptual enhancement of tones by frequency shifts

In a chord of pure tones with a flat spectral profile, one tone can be perceptually enhanced relative to the other tones by the previous presentation of a slightly different chord. "Intensity enhancement" (IE) is obtained when the component tones of the two chords have the same frequencies, but in the first chord the target of enhancement is attenuated relative to the other tones. "Frequency enhancement" (FE) is obtained when both chords have a flat spectral profile, but the target of enhancement shifts in frequency from the first to the second chord. We report here an experiment in which IE and FE were measured using a task requiring the listener to indicate whether or not the second chord included a tone identical to a subsequent probe tone. The results showed that a global attenuation of the first chord relative to the second chord disrupted IE more than FE. This suggests that the mechanisms of IE and FE are not the same. In accordance with this suggestion, computations of the auditory excitation patterns produced by the chords indicate that the mechanism of IE is not sufficient to explain FE for small frequency shifts.

Auditory discrimination of frequency ratios: the octave singularity

Sensitivity to frequency ratios is essential for the perceptual processing of complex sounds and the appreciation of music. This study assessed the effect of ratio simplicity on ratio discrimination for pure tones presented either simultaneously or sequentially. Each stimulus consisted of four 100-ms pure tones, equally spaced in terms of frequency ratio and presented at a low intensity to limit interactions in the auditory periphery. Listeners had to discriminate between a reference frequency ratio of 0.97 octave (about 1.96:1) and target frequency ratios, which were larger than the reference. In the simultaneous condition, the obtained psychometric functions were nonmonotonic: as the target frequency ratio increased from 0.98 octave to 1.04 octaves, discrimination performance initially increased, then decreased, and then increased again; performance was better when the target was exactly one octave (2:1) than when the target was slightly larger. In the sequential condition, by contrast, the psychometric functions were monotonic and there was no effect of frequency ratio simplicity. A control experiment verified that the non-monotonicity observed in the simultaneous condition did not originate from peripheral interactions between the tones. Our results indicate that simultaneous octaves are recognized as "special" frequency intervals by a mechanism that is insensitive to the sign (positive or negative) of deviations from the octave, whereas this is apparently not the case for sequential octaves.

Global not local masker features govern the auditory continuity illusion

When an acoustic signal is temporarily interrupted by another sound, it is sometimes heard as continuing through, even when the signal is actually turned off during the interruption-an effect known as the "auditory continuity illusion." A widespread view is that the illusion can only occur when peripheral neural responses contain no evidence that the signal was interrupted. Here we challenge this view using a combination of psychophysical measures from human listeners and computational simulations with a model of the auditory periphery. The results reveal that the illusion seems to depend more on the overall specific loudness than on the peripheral masking properties of the interrupting sound. This finding indicates that the continuity illusion is determined by the global features, rather than the fine-grained temporal structure, of the interrupting sound, and argues against the view that the illusion arises in the auditory periphery.

Further evidence that fundamental-frequency difference limens measure pitch discrimination

Difference limens for complex tones (DLCs) that differ in F0 are widely regarded as a measure of periodicity-pitch discrimination. However, because F0 changes are inevitably accompanied by changes in the frequencies of the harmonics, DLCs may actually reflect the discriminability of individual components. To test this hypothesis, DLCs were measured for complex tones, the component frequencies of which were shifted coherently upward or downward by ΔF = 0%, 25%, 37.5%, or 50% of the F0, yielding fully harmonic (ΔF = 0%), strongly inharmonic (ΔF = 25%, 37.5%), or odd-harmonic (ΔF = 50%) tones. If DLCs truly reflect periodicity-pitch discriminability, they should be larger (worse) for inharmonic tones than for harmonic and odd harmonic tones because inharmonic tones have a weaker pitch. Consistent with this prediction, the results of two experiments showed a non-monotonic dependence of DLCs on ΔF, with larger DLCs for ΔF's of ± 25% or ± 37.5% than for ΔF's of 0 or ± 50% of F0. These findings are consistent with models of pitch perception that involve harmonic templates or with an autocorrelation-based model provided that more than just the highest peak in the summary autocorrelogram is taken into account.

Intensity-invariant coding in the auditory system

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.

A new model for calculating auditory excitation patterns and loudness for cases of cochlear hearing loss

A model for calculating auditory excitation patterns and loudness for steady sounds for normal hearing is extended to deal with cochlear hearing loss. The filters used in the model have a double ROEX-shape, the gain of the narrow active filter being controlled by the output of the broad passive filter. It is assumed that the hearing loss at each audiometric frequency can be partitioned into a loss due to dysfunction of outer hair cells (OHCs) and a loss due to dysfunction of inner hair cells (IHCs). OHC loss is modeled by decreasing the maximum gain of the active filter, which results in increased absolute threshold, reduced compressive nonlinearity and reduced frequency selectivity. IHC loss is modeled by a level-dependent attenuation of excitation level, which results in elevated absolute threshold. The magnitude of OHC loss and IHC loss can be derived from measures of loudness recruitment and the measured absolute threshold, using an iterative procedure. The model accurately fits loudness recruitment data obtained using subjects with unilateral or highly asymmetric cochlear hearing loss who were required to make loudness matches between tones presented alternately to the two ears. With the same parameters, the model predicted loudness matches between narrowband and broadband sound reasonably well, reflecting loudness summation. The model can also predict when a dead region is present.

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.

Practical ranges of loudness levels of various types of environmental noise, including traffic noise, aircraft noise, and industrial noise

In environmental noise control one commonly employs the A-weighted sound level as an approximate measure of the effect of noise on people. A measure that is more closely related to direct human perception of noise is the loudness level. At constant A-weighted sound level, the loudness level of a noise signal varies considerably with the shape of the frequency spectrum of the noise signal. In particular the bandwidth of the spectrum has a large effect on the loudness level, due to the effect of critical bands in the human hearing system. The low-frequency content of the spectrum also has an effect on the loudness level. In this note the relation between loudness level and A-weighted sound level is analyzed for various environmental noise spectra, including spectra of traffic noise, aircraft noise, and industrial noise. From loudness levels calculated for these environmental noise spectra, diagrams are constructed that show the relation between loudness level, A-weighted sound level, and shape of the spectrum. The diagrams show that the upper limits of the loudness level for broadband environmental noise spectra are about 20 to 40 phon higher than the lower limits for narrowband spectra, which correspond to the loudness levels of pure tones. The diagrams are useful for assessing limitations and potential improvements of environmental noise control methods and policy based on A-weighted sound levels.

The temporal weighting of loudness: effects of the level profile

In four experiments, we studied the influence of the level profile of time-varying sounds on temporal perceptual weights for loudness. The sounds consisted of contiguous wideband noise segments on which independent random-level perturbations were imposed. Experiment 1 showed that in sounds with a flat level profile, the first segment receives the highest weight (primacy effect). If, however, a gradual increase in level (fade-in) was imposed on the first few segments, the temporal weights showed a delayed primacy effect: The first unattenuated segment received the highest weight, while the fade-in segments were virtually ignored. This pattern argues against a capture of attention to the onset as the origin of the primacy effect. Experiment 2 demonstrated that listeners adjust their temporal weights to the level profile on a trial-by-trial basis. Experiment 3 ruled out potentially inferior intensity resolution at lower levels as the cause of the delayed primacy effect. Experiment 4 showed that the weighting patterns cannot be explained by perceptual segmentation of the sounds into a variable and a stable part. The results are interpreted in terms of memory and attention processes. We demonstrate that the prediction of loudness can be improved significantly by allowing for nonuniform temporal weights.

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.

Tinnitus loudness in quiet and noise after resection of vestibular schwannoma

OBJECTIVES

(1) To use a loudness model to assess the influence of loudness recruitment on estimates of the loudness of tinnitus obtained by loudness matching; (2) To compare the effect of background noise on the loudness of tinnitus for individuals who are unilaterally deaf after resection of vestibular schwannoma (VS) and those with idiopathic tinnitus.

BACKGROUND

After translabyrinthine resection of VS, patients experience unilateral deafness and tinnitus in the operated ear. Most complain that their tinnitus is more bothersome in noisy environments, unlike those with idiopathic tinnitus.

PARTICIPANTS

Unilaterally deaf individuals experiencing tinnitus as a consequence of VS surgery and a comparison group with idiopathic tinnitus.

METHODS

Participants adjusted the level of a probe tone at the frequency where their hearing was best to match the loudness of their tinnitus in quiet; for VS participants, matches were made using a probe in the unaffected ear. Matches were then obtained in the presence of threshold-equalizing noise.

RESULTS

For those with idiopathic tinnitus, the probe loudness level, calculated using a loudness model, was almost invariant with hearing loss at the probe frequency and was usually between 20 and 50 phons. For the VS group, the probe loudness level ranged from 6 to 51 phons. With increasing threshold-equalizing-noise level, the loudness match decreased slightly for the comparison group but increased significantly for the VS group.

CONCLUSION

The tinnitus in quiet had a moderate loudness for both groups. Background noise slightly decreased tinnitus loudness for most participants with idiopathic tinnitus but increased tinnitus loudness for VS participants. We propose 2 possible mechanisms for the effect of noise in the VS group.

The loudness of sounds whose spectra differ at the two ears

Moore and Glasberg [(2007). J. Acoust. Soc. Am. 121, 1604-1612] developed a model for predicting the loudness of dichotic sounds. The model gave accurate predictions of data in the literature, except for an experiment of Zwicker and Zwicker [(1991). J. Acoust. Soc. Am. 89, 756-764], in which sounds with non-overlapping spectra were presented to the two ears. The input signal was noise with the same intensity in each critical band (bark). This noise was filtered into 24 bands each 1 bark wide. The bands were then grouped into wider composite bands (consisting of 1, 2, 4, or 12 successive sub-bands) and each composite band was presented either to one ear or the other. Loudness estimates obtained using a scaling procedure decreased somewhat as the number of composite bands increased (and their width decreased), but the predictions of the model showed the opposite pattern. This experiment was similar to that of Zwicker and Zwicker, except that the widths of the bands were based on the ERB(N)-number scale, and a loudness-matching procedure was used. The pattern of the results was consistent with the predictions of the model, showing an increase in loudness as the number of composite bands increased and their spacing decreased.

Modeling speech intelligibility in quiet and noise in listeners with normal and impaired hearing

The speech intelligibility index (SII) is an often used calculation method for estimating the proportion of audible speech in noise. For speech reception thresholds (SRTs), measured in normally hearing listeners using various types of stationary noise, this model predicts a fairly constant speech proportion of about 0.33, necessary for Dutch sentence intelligibility. However, when the SII model is applied for SRTs in quiet, the estimated speech proportions are often higher, and show a larger inter-subject variability, than found for speech in noise near normal speech levels [65 dB sound pressure level (SPL)]. The present model attempts to alleviate this problem by including cochlear compression. It is based on a loudness model for normally hearing and hearing-impaired listeners of Moore and Glasberg [(2004). Hear. Res. 188, 70-88]. It estimates internal excitation levels for speech and noise and then calculates the proportion of speech above noise and threshold using similar spectral weighting as used in the SII. The present model and the standard SII were used to predict SII values in quiet and in stationary noise for normally hearing and hearing-impaired listeners. The present model predicted SIIs for three listener types (normal hearing, noise-induced, and age-induced hearing loss) with markedly less variability than the standard SII.

Spectro-temporal characteristics of speech at high frequencies, and the potential for restoration of audibility to people with mild-to-moderate hearing loss

OBJECTIVES

It is possible for auditory prostheses to provide amplification for frequencies above 6 kHz. However, most current hearing-aid fitting procedures do not give recommended gains for such high frequencies. This study was intended to provide information that could be useful in quantifying appropriate high-frequency gains, and in establishing the population of hearing-impaired people who might benefit from such amplification.

DESIGN

The study had two parts. In the first part, wide-bandwidth recordings of normal conversational speech were obtained from a sample of male and female talkers. The recordings were used to determine the mean spectral shape over a wide frequency range, and to determine the distribution of levels (the speech dynamic range) as a function of center frequency. In the second part, audiometric thresholds were measured for frequencies of 0.125, 0.25, 0.5, 1, 2, 3, 4, 6, 8, 10, and 12.5 kHz for both ears of 31 people selected to have mild or moderate cochlear hearing loss. The hearing loss was never greater than 70 dB for any frequency up to 4 kHz.

RESULTS

The mean spectrum level of the speech fell progressively with increasing center frequency above about 0.5 kHz. For speech with an overall level of 65 dB SPL, the mean 1/3-octave level was 49 and 37 dB SPL for center frequencies of 1 and 10 kHz, respectively. The dynamic range of the speech was similar for center frequencies of 1 and 10 kHz. The part of the dynamic range below the root-mean-square level was larger than reported in previous studies. The mean audiometric thresholds at high frequencies (10 and 12.5 kHz) were relatively high (69 and 77 dB HL, respectively), even though the mean thresholds for frequencies below 4 kHz were 41 dB HL or better.

CONCLUSIONS

To partially restore audibility for a hearing loss of 65 dB at 10 kHz would require an effective insertion gain of about 36 dB at 10 kHz. With this gain, audibility could be (partly) restored for 25 of the 62 ears assessed.

Effects of moderate cochlear hearing loss on the ability to benefit from temporal fine structure information in speech

Speech reception thresholds (SRTs) were measured with a competing talker background for signals processed to contain variable amounts of temporal fine structure (TFS) information, using nine normal-hearing and nine hearing-impaired subjects. Signals (speech and background talker) were bandpass filtered into channels. Channel signals for channel numbers above a "cut-off channel" (CO) were vocoded to remove TFS information, while channel signals for channel numbers of CO and below were left unprocessed. Signals from all channels were combined. As a group, hearing-impaired subjects benefited less than normal-hearing subjects from the additional TFS information that was available as CO increased. The amount of benefit varied between hearing-impaired individuals, with some showing no improvement in SRT and one showing an improvement similar to that for normal-hearing subjects. The reduced ability to take advantage of TFS information in speech may partially explain why subjects with cochlear hearing loss get less benefit from listening in a fluctuating background than normal-hearing subjects. TFS information may be important in identifying the temporal "dips" in such a background.

Modeling binaural loudness

A survey of data on the perception of binaurally presented sounds indicates that loudness summation across ears is less than perfect; a diotic sound is less than twice as loud as the same sound presented monaurally. The loudness model proposed by Moore et al. [J. Audio Eng. Soc. 45, 224-240 (1997)] determines the loudness of binaural stimuli by a simple summation of loudness across ears. It is described here how the model can be modified so as to give more accurate predictions of the loudness of binaurally presented sounds, including cases where the sounds at the two ears differ in level, frequency or both. The modification is based on the idea that there are inhibitory interactions between the internal representations of the signals at the two ears, such that a signal at the left ear inhibits (reduces) the loudness evoked by a signal at the right ear, and vice versa. The inhibition is assumed to spread across frequency channels. The modified model gives reasonably accurate predictions of a variety of data on the loudness of binaural stimuli, including data obtained using loudness scaling and loudness matching procedures.