Time Domain Computation of a Nonlinear Nonlocal Cochlear Model with Applications to Multitone Interaction in Hearing

A nonlinear nonlocal cochlear model of the transmission line type is studied in order to capture the multitone interactions and resulting tonal suppression effects. The model can serve as a module for voice signal processing, it is a one dimensional (in space) damped dispersive nonlinear PDE based on mechanics and phenomenology of hearing. It describes the motion of basilar membrane (BM) in the cochlea driven by input pressure waves. Both elastic damping and selective longitudinal fluid damping are present. The former is nonlinear and nonlocal in BM displacement, and plays a key role in capturing tonal interactions. The latter is active only near the exit boundary (helicotrema), and is built in to damp out the remaining long waves. The initial boundary value problem is numerically solved with a semi-implicit second order finite difference method. Solutions reach a multi-frequency quasi-steady state. Numerical results are shown on two tone suppression from both high-frequency and low-frequency sides, consistent with known behavior of two tone suppression. Suppression effects among three tones are demonstrated by showing how the response magnitudes of the fixed two tones are reduced as we vary the third tone in frequency and amplitude. We observe qualitative agreement of our model solutions with existing cat auditory neural data. The model is thus simple and efficient as a processing tool for voice signals.Comment: 23 pages,7 figures; added reference

A Novel Ergodic Discrete Difference Equation Cochlear Model

In this paper, a novel hardware-efficient electronic circuit cochlear model, the dynamics of which are described by an ergodic cellular automaton, is presented. Based on theoretical and numerical analyses, a parameter setting method so that the presented model properly works as a cochlear model is proposed. It is shown that the presented cochlear model designed by the proposed parameter setting method can reproduce typical nonlinear sound processing functions of mammalian cochleae such as nonlinear compression, two-tone suppression and two-tone distortion products. Furthermore, the presented model is implemented by a field programmable gate array (FPGA) and its operations are validated by experiments. It is shown that the presented model is much more hardware-efficient (i.e., consumes many fewer circuits elements) compared to some other electronic circuit cochlear models

A Biophysical Model of the Role of the Outer Hair Cell in Cochlear Nonlinearity

It has been observed that the response characteristics of the basilar membrane in normal living cochleae are both frequency and level-sensitive (Robles & Ruggero 2001). The quality factor of the tuning curve is large at low sound levels and decreases as the sound level increases, and the peak of the tuning curve moves towards lower frequencies as the sound level increases. The current study proposes a nonlinear cochlear model that responds adaptively to the incoming sounds via feedback control arising from the mechanical attributes of the cochlear partition. These attributes are dependent on the membrane potential of the outer hair cells (He & Dallos 1999, Santos-Sacchi 1992). A parallel resistor-capacitor circuit analogy of the outer hair cell with related perilymph and endolymph potentials is designed to simulate sound-evoked changes in the outer hair cell membrane potential. Nonlinear responses of the cochlea, such as compression and two tone suppression, can be explained using this model. Furthermore, it has been shown that the basilar membrane response to pure tone stimuli is attenuated by directly stimulating the medial olivo-cochlear bundle using electrical shocks (Cooper & Guinan 2006). Basilar membrane responses in the presence of efferent stimulation can be demonstrated using the same model, through modulation of the outer hair cell rnembrane potential. The proposed model provides a unified account of the combined effect of sounds and efferent stimulation on cochlear responses

Synchronization of a Nonlinear Oscillator: Processing the Cf Component of the Echo-Response Signal in the Cochlea of the Mustached Bat

Cochlear microphonic potential (CM) was recorded from the CF2 region and the sparsely innervated zone (the mustached bat's cochlea fovea) that is specialized for analyzing the Doppler-shifted echoes of the first-harmonic (~61 kHz) of the constant-frequency component of the echolocation call. Temporal analysis of the CM, which is tuned sharply to the 61 kHz cochlear resonance, revealed that at the resonance frequency, and within 1 msec of tone onset, CM is broadly tuned with linear magnitude level functions. CM measured during the ongoing tone and in the ringing after tone offset is 50 dB more sensitive, is sharply tuned, has compressive level functions, and the phase leads onset CM by 90°: an indication that cochlear responses are amplified during maximum basilar membrane velocity. For high-level tones above the resonance frequency, CM appears at tone onset and after tone offset. Measurements indicate that the two oscillators responsible for the cochlear resonance, presumably the basilar and tectorial membranes, move together in phase during the ongoing tone, thereby minimizing net shear between them and hair cell excitation. For tones within 2 kHz of the cochlear resonance the frequency of CM measured within 2 msec of tone onset is not that of the stimulus but is proportional to it. For tones just below the cochlear resonance region CM frequency is a constant amount below that of the stimulus depending on CM measurement delay from tone onset. The frequency responses of the CM recorded from the cochlear fovea can be accounted for through synchronization between the nonlinear oscillators responsible for the cochlear resonance and the stimulus tone

Effects of noise suppression and envelope dynamic range compression on the intelligibility of vocoded sentences for a tonal language

Vocoder simulation studies have suggested that the carrier signal type employed affects the intelligibility of vocoded speech. The present work further assessed how carrier signal type interacts with additional signal processing, namely, single-channel noise suppression and envelope dynamic range compression, in determining the intelligibility of vocoder simulations. In Experiment 1, Mandarin sentences that had been corrupted by speech spectrum-shaped noise (SSN) or two-talker babble (2TB) were processed by one of four single-channel noise-suppression algorithms before undergoing tone-vocoded (TV) or noise-vocoded (NV) processing. In Experiment 2, dynamic ranges of multiband envelope waveforms were compressed by scaling of the mean-removed envelope waveforms with a compression factor before undergoing TV or NV processing. TV Mandarin sentences yielded higher intelligibility scores with normal-hearing (NH) listeners than did noise-vocoded sentences. The intelligibility advantage of noise-suppressed vocoded speech depended on the masker type (SSN vs 2TB). NV speech was more negatively influenced by envelope dynamic range compression than was TV speech. These findings suggest that an interactional effect exists between the carrier signal type employed in the vocoding process and envelope distortion caused by signal processing

A frequency-selective feedback model of auditory efferent suppression and its implications for the recognition of speech in noise

The potential contribution of the peripheral auditory efferent system to our understanding of speech in a background of competing noise was studied using a computer model of the auditory periphery and assessed using an automatic speech recognition system. A previous study had shown that a fixed efferent attenuation applied to all channels of a multi-channel model could improve the recognition of connected digit triplets in noise [G. J. Brown, R. T. Ferry, and R. Meddis, J. Acoust. Soc. Am. 127, 943?954 (2010)]. In the current study an anatomically justified feedback loop was used to automatically regulate separate attenuation values for each auditory channel. This arrangement resulted in a further enhancement of speech recognition over fixed-attenuation conditions. Comparisons between multi-talker babble and pink noise interference conditions suggest that the benefit originates from the model?s ability to modify the amount of suppression in each channel separately according to the spectral shape of the interfering sounds

Determination and evaluation of clinically efficient stopping criteria for the multiple auditory steady-state response technique

Background: Although the auditory steady-state response (ASSR) technique utilizes objective statistical detection algorithms to estimate behavioural hearing thresholds, the audiologist still has to decide when to terminate ASSR recordings introducing once more a certain degree of subjectivity. Aims: The present study aimed at establishing clinically efficient stopping criteria for a multiple 80-Hz ASSR system. Methods: In Experiment 1, data of 31 normal hearing subjects were analyzed off-line to propose stopping rules. Consequently, ASSR recordings will be stopped when (1) all 8 responses reach significance and significance can be maintained for 8 consecutive sweeps; (2) the mean noise levels were ≤ 4 nV (if at this “≤ 4-nV” criterion, p-values were between 0.05 and 0.1, measurements were extended only once by 8 sweeps); and (3) a maximum amount of 48 sweeps was attained. In Experiment 2, these stopping criteria were applied on 10 normal hearing and 10 hearing-impaired adults to asses the efficiency. Results: The application of these stopping rules resulted in ASSR threshold values that were comparable to other multiple-ASSR research with normal hearing and hearing-impaired adults. Furthermore, in 80% of the cases, ASSR thresholds could be obtained within a time-frame of 1 hour. Investigating the significant response-amplitudes of the hearing-impaired adults through cumulative curves indicated that probably a higher noise-stop criterion than “≤ 4 nV” can be used. Conclusions: The proposed stopping rules can be used in adults to determine accurate ASSR thresholds within an acceptable time-frame of about 1 hour. However, additional research with infants and adults with varying degrees and configurations of hearing loss is needed to optimize these criteria