Development of an Arabic "Command in Noise" Hearing Test to Assess Fitness for Duty.

Objective: The goal is to implement the developed speech material in a hearing test to assess auditory fitness for duty (AFFD), specifically in areas where the intelligibility of spoken commands is essential. Design: In study 1, a speech corpus with equal intelligibility was constructed using constant stimuli to test each target word's psychometric functions. Study 2 used an adaptive interleaving procedure to maximize equalized terms. Study 3 used Monte Carlo simulations to determine speech test accuracy. Study sample: Study 1 (n = 24) and study 2 (n = 20) were completed by civilians with normal hearing. Study 3 ran 10,000 simulations per condition across various conditions varying in slopes and speech recognition thresholds (SRTs). Results: Studies 1 and 2 produced three 8-word wordlists. The mean, standard deviation in dB SNR is -13.1 1.2 for wordlist 1, -13.7 1.6 for wordlist 2, and -13.7 1.3 for wordlist 3, with word SRTs within 3.4 dB SNR. Study 3 revealed that a 6 dB SNR range is appropriate for equally understandable speech using a closed-set adaptive technique. Conclusion: The developed speech corpus may be used in an AFFD measure. Concerning the homogeneity of the speech in noise test material, care should be taken when generalizing and using ranges and standard deviations from multiple tests.

Investigation of an MAA Test With Virtual Sound Synthesis.

The ability to localize a sound source is very important in our daily life, specifically to analyze auditory scenes in complex acoustic environments. The concept of minimum audible angle (MAA), which is defined as the smallest detectable difference between the incident directions of two sound sources, has been widely used in the research fields of auditory perception to measure localization ability. Measuring MAAs usually involves a reference sound source and either a large number of loudspeakers or a movable sound source in order to reproduce sound sources at a large number of predefined incident directions. However, existing MAA test systems are often cumbersome because they require a large number of loudspeakers or a mechanical rail slide and thus are expensive and inconvenient to use. This study investigates a novel MAA test method using virtual sound source synthesis and avoiding the problems with traditional methods. We compare the perceptual localization acuity of sound sources in two experimental designs: using the virtual presentation and real sound sources. The virtual sound source is reproduced through a pair of loudspeakers weighted by vector-based amplitude panning (VBAP). Results show that the average measured MAA at 0° azimuth is 1.1° and the average measured MAA at 90° azimuth is 3.1° in a virtual acoustic system, meanwhile the average measured MAA at 0° azimuth is about 1.2° and the average measured MAA at 90° azimuth is 3.3° when using the real sound sources. The measurements of the two methods have no significant difference. We conclude that the proposed MAA test system is a suitable alternative to more complicated and expensive setups.

Tolerable delay for speech production and perception: effects of hearing ability and experience with hearing aids.

OBJECTIVE: Processing delay is one of the important factors that limit the development of novel algorithms for hearing devices. In this study, both normal-hearing listeners and listeners with hearing loss were tested for their tolerance of processing delay up to 50 ms using a real-time setup for own-voice and external-voice conditions based on linear processing to avoid confounding effects of time-dependent gain. DESIGN: Participants rated their perceived subjective annoyance for each condition on a 7-point Likert scale. STUDY SAMPLE: Twenty normal-hearing participants and twenty participants with a range of mild to moderate hearing losses. RESULTS: Delay tolerance was significantly greater for the participants with hearing loss in two out of three voice conditions. The average slopes of annoyance ratings were negatively correlated with the degree of hearing loss across participants. A small trend of higher tolerance of delay by experienced users of hearing aids in comparison to new users was not significant. CONCLUSION: The increased tolerance of processing delay for speech production and perception with hearing loss and reduced sensitivity to changes in delay with stronger hearing loss may be beneficial for novel algorithms for hearing devices but the setup used in this study differed from commercial hearing aids.

Auditory inspired machine learning techniques can improve speech intelligibility and quality for hearing-impaired listeners.

Machine-learning based approaches to speech enhancement have recently shown great promise for improving speech intelligibility for hearing-impaired listeners. Here, the performance of three machine-learning algorithms and one classical algorithm, Wiener filtering, was compared. Two algorithms based on neural networks were examined, one using a previously reported feature set and one using a feature set derived from an auditory model. The third machine-learning approach was a dictionary-based sparse-coding algorithm. Speech intelligibility and quality scores were obtained for participants with mild-to-moderate hearing impairments listening to sentences in speech-shaped noise and multi-talker babble following processing with the algorithms. Intelligibility and quality scores were significantly improved by each of the three machine-learning approaches, but not by the classical approach. The largest improvements for both speech intelligibility and quality were found by implementing a neural network using the feature set based on auditory modeling. Furthermore, neural network based techniques appeared more promising than dictionary-based, sparse coding in terms of performance and ease of implementation.

Speech enhancement based on neural networks improves speech intelligibility in noise for cochlear implant users.

Speech understanding in noisy environments is still one of the major challenges for cochlear implant (CI) users in everyday life. We evaluated a speech enhancement algorithm based on neural networks (NNSE) for improving speech intelligibility in noise for CI users. The algorithm decomposes the noisy speech signal into time-frequency units, extracts a set of auditory-inspired features and feeds them to the neural network to produce an estimation of which frequency channels contain more perceptually important information (higher signal-to-noise ratio, SNR). This estimate is used to attenuate noise-dominated and retain speech-dominated CI channels for electrical stimulation, as in traditional n-of-m CI coding strategies. The proposed algorithm was evaluated by measuring the speech-in-noise performance of 14 CI users using three types of background noise. Two NNSE algorithms were compared: a speaker-dependent algorithm, that was trained on the target speaker used for testing, and a speaker-independent algorithm, that was trained on different speakers. Significant improvements in the intelligibility of speech in stationary and fluctuating noises were found relative to the unprocessed condition for the speaker-dependent algorithm in all noise types and for the speaker-independent algorithm in 2 out of 3 noise types. The NNSE algorithms used noise-specific neural networks that generalized to novel segments of the same noise type and worked over a range of SNRs. The proposed algorithm has the potential to improve the intelligibility of speech in noise for CI users while meeting the requirements of low computational complexity and processing delay for application in CI devices.

Enhancement of forward suppression begins in the ventral cochlear nucleus.

A neuron׳s response to a sound can be suppressed by the presentation of a preceding sound. It has been suggested that this suppression is a direct correlate of the psychophysical phenomenon of forward masking, however, forward suppression, as measured in the responses of the auditory nerve, was insufficient to account for behavioural performance. In contrast the neural suppression seen in the inferior colliculus and auditory cortex was much closer to psychophysical performance. In anaesthetised guinea-pigs, using a physiological two-interval forced-choice threshold tracking algorithm to estimate suppressed (masked) thresholds, we examine whether the enhancement of suppression can occur at an earlier stage of the auditory pathway, the ventral cochlear nucleus (VCN). We also compare these responses with the responses from the central nucleus of the inferior colliculus (ICc) using the same preparation. In both nuclei, onset-type neurons showed the greatest amounts of suppression (16.9-33.5dB) and, in the VCN, these recovered with the fastest time constants (14.1-19.9ms). Neurons with sustained discharge demonstrated reduced masking (8.9-12.1dB) and recovery time constants of 27.2-55.6ms. In the VCN the decrease in growth of suppression with increasing suppressor level was largest for chopper units and smallest for onset-type units. The threshold elevations recorded for most unit types are insufficient to account for the magnitude of forward masking as measured behaviourally, however, onset responders, in both the cochlear nucleus and inferior colliculus demonstrate a wide dynamic range of suppression, similar to that observed in human psychophysics.

Speech quality evaluation of a sparse coding shrinkage noise reduction algorithm with normal hearing and hearing impaired listeners.

Although there are numerous papers describing single-channel noise reduction strategies to improve speech perception in a noisy environment, few studies have comprehensively evaluated the effects of noise reduction algorithms on speech quality for hearing impaired (HI). A model-based sparse coding shrinkage (SCS) algorithm has been developed, and has shown previously (Sang et al., 2014) that it is as competitive as a state-of-the-art Wiener filter approach in speech intelligibility. Here, the analysis is extended to include subjective quality ratings and a method called Interpolated Paired Comparison Rating (IPCR) is adopted to quantitatively link the benefit of speech intelligibility and speech quality. The subjective quality tests are performed through IPCR to efficiently quantify noise reduction effects on speech quality. Objective measures including frequency-weighted segmental signal-to-noise ratio (fwsegSNR), perceptual evaluation of speech quality (PESQ) and hearing aid speech quality index (HASQI) are adopted to predict the noise reduction effects. Results show little difference in speech quality between the SCS and the Wiener filter algorithm but a difference in quality rating between the HI and NH listeners. HI listeners generally gave better quality ratings of noise reduction algorithms than NH listeners. However, SCS reduced the noise more efficiently at the cost of higher distortions that were detected by NH but not by the HI. SCS is a promising candidate for noise reduction algorithms for HI. In general, care needs to be taken when adopting algorithms that were originally developed for NH participants into hearing aid applications. An algorithm that is evaluated negatively with NH might still bring benefits for HI participants.

Sparse Nonnegative Matrix Factorization Strategy for Cochlear Implants.

Current cochlear implant (CI) strategies carry speech information via the waveform envelope in frequency subbands. CIs require efficient speech processing to maximize information transfer to the brain, especially in background noise, where the speech envelope is not robust to noise interference. In such conditions, the envelope, after decomposition into frequency bands, may be enhanced by sparse transformations, such as nonnegative matrix factorization (NMF). Here, a novel CI processing algorithm is described, which works by applying NMF to the envelope matrix (envelopogram) of 22 frequency channels in order to improve performance in noisy environments. It is evaluated for speech in eight-talker babble noise. The critical sparsity constraint parameter was first tuned using objective measures and then evaluated with subjective speech perception experiments for both normal hearing and CI subjects. Results from vocoder simulations with 10 normal hearing subjects showed that the algorithm significantly enhances speech intelligibility with the selected sparsity constraints. Results from eight CI subjects showed no significant overall improvement compared with the standard advanced combination encoder algorithm, but a trend toward improvement of word identification of about 10 percentage points at +15 dB signal-to-noise ratio (SNR) was observed in the eight CI subjects. Additionally, a considerable reduction of the spread of speech perception performance from 40% to 93% for advanced combination encoder to 80% to 100% for the suggested NMF coding strategy was observed.

Sensitivity to Envelope Interaural Time Differences at High Modulation Rates.

Sensitivity to interaural time differences (ITDs) conveyed in the temporal fine structure of low-frequency tones and the modulated envelopes of high-frequency sounds are considered comparable, particularly for envelopes shaped to transmit similar fidelity of temporal information normally present for low-frequency sounds. Nevertheless, discrimination performance for envelope modulation rates above a few hundred Hertz is reported to be poor-to the point of discrimination thresholds being unattainable-compared with the much higher (>1,000 Hz) limit for low-frequency ITD sensitivity, suggesting the presence of a low-pass filter in the envelope domain. Further, performance for identical modulation rates appears to decline with increasing carrier frequency, supporting the view that the low-pass characteristics observed for envelope ITD processing is carrier-frequency dependent. Here, we assessed listeners' sensitivity to ITDs conveyed in pure tones and in the modulated envelopes of high-frequency tones. ITD discrimination for the modulated high-frequency tones was measured as a function of both modulation rate and carrier frequency. Some well-trained listeners appear able to discriminate ITDs extremely well, even at modulation rates well beyond 500 Hz, for 4-kHz carriers. For one listener, thresholds were even obtained for a modulation rate of 800 Hz. The highest modulation rate for which thresholds could be obtained declined with increasing carrier frequency for all listeners. At 10 kHz, the highest modulation rate at which thresholds could be obtained was 600 Hz. The upper limit of sensitivity to ITDs conveyed in the envelope of high-frequency modulated sounds appears to be higher than previously considered.

Evaluation of the sparse coding shrinkage noise reduction algorithm in normal hearing and hearing impaired listeners.

Although there are numerous single-channel noise reduction strategies to improve speech perception in noise, most of them improve speech quality but do not improve speech intelligibility, in circumstances where the noise and speech have similar frequency spectra. Current exceptions that may improve speech intelligibility are those that require a priori knowledge of the speech or noise statistics, which limits practical application. Hearing impaired (HI) listeners suffer more in speech intelligibility than normal hearing listeners (NH) in the same noisy environment, so developing better single-channel noise reduction algorithms for HI listeners is justified. Our model-based "sparse coding shrinkage" (SCS) algorithm extracts key speech information in noisy speech. We evaluate it by comparison with a state-of-the-art Wiener filtering approach using speech intelligibility tests with NH and HI listeners. The model-based SCS algorithm relies only on statistical signal information without prior information. Results show that the SCS algorithm improves speech intelligibility in stationary noise and is comparable to the Wiener filtering algorithm. Both algorithms improve intelligibility for HI listeners but not for NH listeners. Improvement is less in fluctuating (babble) noise than in stationary noise. Both noise reduction algorithms perform better at higher input signal-to-noise ratios (SNR) where HI listeners can benefit but where NH listeners have already reached ceiling performance. The difference between NH and HI subjects in intelligibility gain depends fundamentally on the input SNR rather than the hearing loss level. We conclude that HI listeners need different signal processing algorithms from NH subjects and that the SCS algorithm offers a promising alternative to Wiener filtering. Performance of all noise reduction algorithms is likely to vary according to extent of hearing loss and algorithms that show little benefit for listeners with moderate hearing loss may be more beneficial for listeners with more severe hearing loss.

Development of a real time sparse non-negative matrix factorization module for cochlear implants by using xPC target.

Cochlear implants (CIs) require efficient speech processing to maximize information transmission to the brain, especially in noise. A novel CI processing strategy was proposed in our previous studies, in which sparsity-constrained non-negative matrix factorization (NMF) was applied to the envelope matrix in order to improve the CI performance in noisy environments. It showed that the algorithm needs to be adaptive, rather than fixed, in order to adjust to acoustical conditions and individual characteristics. Here, we explore the benefit of a system that allows the user to adjust the signal processing in real time according to their individual listening needs and their individual hearing capabilities. In this system, which is based on MATLAB®, SIMULINK® and the xPC Target™ environment, the input/outupt (I/O) boards are interfaced between the SIMULINK blocks and the CI stimulation system, such that the output can be controlled successfully in the manner of a hardware-in-the-loop (HIL) simulation, hence offering a convenient way to implement a real time signal processing module that does not require any low level language. The sparsity constrained parameter of the algorithm was adapted online subjectively during an experiment with normal-hearing subjects and noise vocoded speech simulation. Results show that subjects chose different parameter values according to their own intelligibility preferences, indicating that adaptive real time algorithms are beneficial to fully explore subjective preferences. We conclude that the adaptive real time systems are beneficial for the experimental design, and such systems allow one to conduct psychophysical experiments with high ecological validity.

Semi-automatic attenuation of cochlear implant artifacts for the evaluation of late auditory evoked potentials.

Electrical artifacts caused by the cochlear implant (CI) contaminate electroencephalographic (EEG) recordings from implanted individuals and corrupt auditory evoked potentials (AEPs). Independent component analysis (ICA) is efficient in attenuating the electrical CI artifact and AEPs can be successfully reconstructed. However the manual selection of CI artifact related independent components (ICs) obtained with ICA is unsatisfactory, since it contains expert-choices and is time consuming. We developed a new procedure to evaluate temporal and topographical properties of ICs and semi-automatically select those components representing electrical CI artifact. The CI Artifact Correction (CIAC) algorithm was tested on EEG data from two different studies. The first consists of published datasets from 18 CI users listening to environmental sounds. Compared to the manual IC selection performed by an expert the sensitivity of CIAC was 91.7% and the specificity 92.3%. After CIAC-based attenuation of CI artifacts, a high correlation between age and N1-P2 peak-to-peak amplitude was observed in the AEPs, replicating previously reported findings and further confirming the algorithm's validity. In the second study AEPs in response to pure tone and white noise stimuli from 12 CI users that had also participated in the other study were evaluated. CI artifacts were attenuated based on the IC selection performed semi-automatically by CIAC and manually by one expert. Again, a correlation between N1 amplitude and age was found. Moreover, a high test-retest reliability for AEP N1 amplitudes and latencies suggested that CIAC-based attenuation reliably preserves plausible individual response characteristics. We conclude that CIAC enables the objective and efficient attenuation of the CI artifact in EEG recordings, as it provided a reasonable reconstruction of individual AEPs. The systematic pattern of individual differences in N1 amplitudes and latencies observed with different stimuli at different sessions, strongly suggests that CIAC can overcome the electrical artifact problem. Thus CIAC facilitates the use of cortical AEPs as an objective measurement of auditory rehabilitation.

Relationship between speech recognition in noise and sparseness.

OBJECTIVE: Established methods for predicting speech recognition in noise require knowledge of clean speech signals, placing limitations on their application. The study evaluates an alternative approach based on characteristics of noisy speech, specifically its sparseness as represented by the statistic kurtosis. DESIGN: Experiments 1 and 2 involved acoustic analysis of vowel-consonant-vowel (VCV) syllables in babble noise, comparing kurtosis, glimpsing areas, and extended speech intelligibility index (ESII) of noisy speech signals with one another and with pre-existing speech recognition scores. Experiment 3 manipulated kurtosis of VCV syllables and investigated effects on speech recognition scores in normal-hearing listeners. STUDY SAMPLE: Pre-existing speech recognition data for Experiments 1 and 2; seven normal-hearing participants for Experiment 3. RESULTS: Experiments 1 and 2 demonstrated that kurtosis calculated in the time-domain from noisy speech is highly correlated (r > 0.98) with established prediction models: glimpsing and ESII. All three measures predicted speech recognition scores well. The final experiment showed a clear monotonic relationship between speech recognition scores and kurtosis. CONCLUSIONS: Speech recognition performance in noise is closely related to the sparseness (kurtosis) of the noisy speech signal, at least for the types of speech and noise used here and for listeners with normal hearing.

Uncovering auditory evoked potentials from cochlear implant users with independent component analysis.

Auditory evoked potentials (AEPs) provide an objective measure of auditory cortical function, but AEPs from cochlear implant (CI) users are contaminated by an electrical artifact. Here, we investigated the effects of electrical artifact attenuation on AEP quality. The ability of independent component analysis (ICA) in attenuating the CI artifact while preserving the AEPs was evaluated. AEPs recovered from CI users were systematically correlated with age, demonstrating that individual differences were well preserved. CI users with high-quality AEPs were characterized by a significantly shorter duration of deafness. Finally, a simulation study revealed very high spatial correlations between original and recovered normal hearing AEPs (r>.95) that were previously contaminated with CI artifacts. The results confirm that after ICA, good quality AEPs can be recovered, facilitating the objective, noninvasive study of auditory cortex function in CI users.

Behavioral and physiological correlates of temporal pitch perception in electric and acoustic hearing.

In the "4-6" condition of experiment 1, normal-hearing (NH) listeners compared the pitch of a bandpass-filtered pulse train, whose inter-pulse intervals (IPIs) alternated between 4 and 6 ms, to that of isochronous pulse trains. Consistent with previous results obtained at a lower signal level, the pitch of the 4-6 stimulus corresponded to that of an isochronous pulse train having a period of 5.7 ms-longer than the mean IPI of 5 ms. In other conditions the IPI alternated between 3.5-5.5 and 4.5-6.5 ms. Experiment 2 was similar but presented electric pulse trains to one channel of a cochlear implant. In both cases, as overall IPI increased, the pitch of the alternating-interval stimulus approached that of an isochronous train having a period equal to the mean IPI. Experiment 3 measured compound action potentials (CAPs) to alternating-interval stimuli in guinea pigs and in NH listeners. The CAPs to pulses occurring after 4-ms intervals were smaller than responses to pulses occurring after 6-ms intervals, resulting in a modulated pattern that was independent of overall level. The results are compared to the predictions of a simple model incorporating auditory-nerve (AN) refractoriness, and where pitch is estimated from first-order intervals in the AN response.

Source localization of auditory evoked potentials after cochlear implantation.

Little is known about how the auditory cortex adapts to artificial input as provided by a cochlear implant (CI). We report the case of a 71-year-old profoundly deaf man, who has successfully used a unilateral CI for 4 years. Independent component analysis (ICA) of 61-channel EEG recordings could separate CI-related artifacts from auditory-evoked potentials (AEPs), even though it was the perfectly time-locked CI stimulation that caused the AEPs. AEP dipole source localization revealed contralaterally larger amplitudes in the P1-N1 range, similar to normal hearing individuals. In contrast to normal hearing individuals, the man with the CI showed a 20-ms shorter N1 latency ipsilaterally. We conclude that ICA allows the detailed study of AEPs in CI users.

Contralateral inhibitory and excitatory frequency response maps in the mammalian cochlear nucleus.

There is increasing evidence that the responses of single units in the mammalian cochlear nucleus can be altered by the presentation of contralateral stimuli, although the functional significance of this binaural responsiveness is unknown. To further our understanding of this phenomenon we recorded single-unit (n = 110) response maps from the cochlear nucleus (ventral and dorsal divisions) of the anaesthetized guinea pig in response to presentation of ipsilateral and contralateral pure tones. Many neurones showed no evidence of input from the contralateral ear (n = 41) but other neurones from both ventral and dorsal cochlear nucleus showed clear evidence of contralateral inhibitory input (n = 61). Inhibitory response patterns were divided into two groups. In 36 neurones, contralateral tone-evoked inhibition was closely aligned with the ipsilateral excitatory response map (+/- 0.33 octaves) often extending to low stimulus levels. In 25 neurones, higher threshold contralateral inhibitory responses were found, mostly centred at frequencies greater than 0.33 octaves below the ipsilateral excitation. A few neurones (n = 8) exhibited responses consistent with excitatory input from the contralateral ear, which was closely aligned with the ipsilateral excitation, and were found exclusively in the dorsal cochlear nucleus. The latency of the contralateral interaction was, on average, longer than the ipsilateral latency. Interaural level difference curves are similar to other reports from the cochlear nucleus. Our results are consistent with the idea that contralateral interactions arise from a variety of direct and indirect neuronal projections.

The time course of recovery from suppression and facilitation from single units in the mammalian cochlear nucleus.

The responses to two identical, consecutive pure tone stimuli with varying inter-stimulus intervals (delta ts) were measured for 89 neurons in the cochlear nucleus of the anaesthetised guinea pig. We observed two main effects; either a decrease (suppression) or an increase (facilitation) in response to the second tone followed by an exponential recovery. Response behaviour correlated with the unit type; primary-like, primary-like with notch and transient-chopper units showed a recovery from suppression that was very similar to that already reported in the auditory nerve. For chopper units the strength of the adaptation was correlated with the units regularity of spike discharge; sustained chopper (CS) units showed less suppression than transient choppers. Onset units showed complete suppression at short delta ts. Pause/Build (PB) units responded with increased activity to the second tone. In contrast to previous studies in the cochlear nucleus the recovery from suppression or facilitation was well described by a single exponential function, enabling us to define a recovery time constant and a maximum suppression/facilitation. There appeared to be a hierarchy in the time constant of recovery with PB and CS units showing the longest recovery times and onset units showing the shortest.

The effect of cross-channel synchrony on the perception of temporal regularity.

Temporal models of pitch are based on the assumption that the auditory system measures the time intervals between neural events, and that pitch corresponds to the most common time interval. The current experiments were designed to test whether time intervals are analyzed independently in each peripheral channel, or whether the time-interval analysis in one channel is affected by synchronous activity in other channels. Regular and irregular click trains were filtered into narrow frequency bands to produce target and flanker stimuli. The threshold for discriminating a regular target from an irregular distracter click train was measured in the presence of an irregular masker click train in the target band, as a function of the frequency separation between the target band and a flanker band. The flanker click train was either regular or irregular. The threshold for detecting the regular target was 5-7 dB lower when the flanker was regular. The data indicate that the detection of temporal regularity (and thus, pitch) involves cross-channel processes that can operate over widely separated channels. Model simulations suggest that these cross-channel processes occur after the time-interval extraction stage and that they depend on the similarity, or consistency, of the time-interval patterns in the relevant channels.

Structural and functional asymmetry of lateral Heschl's gyrus reflects pitch perception preference.

The relative pitch of harmonic complex sounds, such as instrumental sounds, may be perceived by decoding either the fundamental pitch (f0) or the spectral pitch (fSP) of the stimuli. We classified a large cohort of 420 subjects including symphony orchestra musicians to be either f0 or fSP listeners, depending on the dominant perceptual mode. In a subgroup of 87 subjects, MRI (magnetic resonance imaging) and magnetoencephalography studies demonstrated a strong neural basis for both types of pitch perception irrespective of musical aptitude. Compared with f0 listeners, fSP listeners possessed a pronounced rightward, rather than leftward, asymmetry of gray matter volume and P50m activity within the pitch-sensitive lateral Heschl's gyrus. Our data link relative hemispheric lateralization with perceptual stimulus properties, whereas the absolute size of the Heschl's gyrus depends on musical aptitude.