Lowel P. O’Mard

Bio: Lowel P. O’Mard is an academic researcher from University of Essex. The author has contributed to research in topics: Basilar membrane & Autocorrelation. The author has an hindex of 8, co-authored 9 publications receiving 1082 citations.

A unitary model of pitch perception

Abstract: A model of the mechanism of residue pitch perception is revisited. It is evaluated in the context of some new empirical results, and it is proposed that the model is able to reconcile a number of differing approaches in the history of theories of pitch perception. The model consists of four sequential processing stages: peripheral frequency selectivity, within-channel half-wave rectification and low-pass filtering, within-channel periodicity extraction, and cross-channel aggregation of the output. The pitch percept is represented by the aggregated periodicity function. Using autocorrelation as the periodicity extraction method and the summary autocorrelation function (SACF) as the method for representing pitch information, it is shown that the model can simulate new experimental results that show how the quality of the pitch percept is influenced by the resolvability of the harmonic components of the stimulus complex. These include: (i) the pitch of harmonic stimuli whose components alternate in phase; (ii) the increased frequency difference limen of tones consisting of higher harmonics; and (iii) the influence of a mistuned harmonic on the pitch of the complex as a function of its harmonic number. To accommodate these paradigms, it was necessary to compare stimuli along the length of the SACF rather than relying upon the highest peak alone. These new results demonstrate that the model responds differently to complexes consisting of low and high harmonics. As a consequence, it is not necessary to postulate two separate mechanisms to explain different pitch percepts associated with resolved and unresolved harmonics.

A revised model of the inner-hair cell and auditory-nerve complex.

Abstract: A revised computational model of the inner-hair cell (IHC) and auditory-nerve (AN) complex is presented and evaluated. Building on previous models, the algorithm is intended as a component for use in more comprehensive models of the auditory periphery. It combines smaller components that aim to be faithful to physiology in so far as is practicable and known. Transduction between cochlear mechanical motion and IHC receptor potential (RP) is simulated using a modification of an existing biophysical IHC model. Changes in RP control the opening of calcium ion channels near the synapse, and local calcium levels determine the probability of the release of neurotransmitter. AN adaptation results from transmitter depletion. The exact timing of AN action potentials is determined by the quantal and stochastic release of neurotransmitter into the cleft. The model reproduces a wide range of animal RP and AN observations. When the input to the model is taken from a suitably nonlinear simulation of the motion of the cochlear partition, the new algorithm is able to simulate the rate-intensity functions of low-, medium-, and high-spontaneous rate AN fibers in response to stimulation both at best frequency and at other frequencies. The variation in fiber type arises in large part from the manipulation of a single parameter in the model: maximum calcium conductance. The model also reproduces quantitatively phase-locking characteristics, relative refractory effects, mean-to-variance ratio, and first- and second-order discharge history effects.

A computational algorithm for computing nonlinear auditory frequency selectivity

Abstract: Computational algorithms that mimic the response of the basilar membrane must be capable of reproducing a range of complex features that are characteristic of the animal observations. These include complex input output functions that are nonlinear near the site's best frequency, but linear elsewhere. This nonlinearity is critical when using the output of the algorithm as the input to models of inner hair cell function and subsequent auditory-nerve models of low- and high-spontaneous rate fibers. We present an algorithm that uses two processing units operating in parallel: one linear and the other compressively nonlinear. The output from the algorithm is the sum of the outputs of the linear and nonlinear processing units. Input to the algorithm is stapes motion and output represents basilar membrane motion. The algorithm is evaluated against published chinchilla and guinea pig observations of basilar membrane and Reissner's membrane motion made using laser velocimetry. The algorithm simulates both quantitatively and qualitatively, differences in input/output functions among three different sites along the cochlear partition. It also simulates quantitatively and qualitatively a range of phenomena including isovelocity functions, phase response, two-tone suppression, impulse response, and distortion products. The algorithm is potentially suitable for development as a bank of filters, for use in more comprehensive models of the peripheral auditory system.

Adaptation in a revised inner-hair cell model.

Abstract: A revised computational model of the inner-hair cell (IHC) and auditory-nerve (AN) complex was recently presented [Sumner et al., J. Acoust. Soc. Am. 111, 2178–2188 (2002)]. One key improvement is that the model reproduces the rate-intensity functions of low- (LSR), medium- (MSR), and high-spontaneous rate (HSR) fibers in the guinea-pig. Here we describe the adaptation characteristics of the model, and how they vary with model fiber type. Adaptation of the revised model for a HSR fiber is in line with an earlier version of the model [Meddis and Hewitt, J. Acoust. Soc. Am. 90, 904–917 (1991)]. In guinea-pig, poststimulus time histograms (PSTH) have been found to show less adaptation in LSR fibers. Evidence from chinchilla suggests that this is due to chronic adaptation resulting from short interstimulus intervals, and that fully recovered LSR fibers actually show more adaptation. However, the model is able to account for both variations of PSTH shape when fully recovered from adaptation. Interstimulus interval can also affect recovery in the model. The model is further tested against data previously used to evaluate models of AN adaptation. The tests are (i) recovery from adaptation of spontaneous rate and (ii) the recovery of response to acoustic stimuli (“forward masking”), (iii) the response to stimulus increments and (iv) decrements, and (v) the conservation of transient components. A HSR model fiber performs similarly to the earlier version of the model. However, there is considerable variation in response to increments and decrements between different model fibers.

Virtual pitch in a computational physiological model

Abstract: A computational model of nervous activity in the auditory nerve, cochlear nucleus, and inferior colliculus is presented and evaluated in terms of its ability to simulate psychophysically-measured pitch perception. The model has a similar architecture to previous autocorrelation models except that the mathematical operations of autocorrelation are replaced by the combined action of thousands of physiologically plausible neuronal components. The evaluation employs pitch stimuli including complex tones with a missing fundamental frequency, tones with alternating phase, inharmonic tones with equally spaced frequencies and iterated rippled noise. Particular attention is paid to differences in response to resolved and unresolved component harmonics. The results indicate that the model is able to simulate qualitatively the related pitch-perceptions. This physiological model is similar in many respects to autocorrelation models of pitch and the success of the evaluations suggests that autocorrelation models may, ...

The Organization of the Cochlear Receptor

Abstract: The yield of alkenols and cycloalkenols is substantially improved by carrying out the reaction of olefins with formaldehyde in the presence of selected catalysts. In accordance with one embodiment, alk-3-en-1-ols are produced in good yields from isobutylene and formaldehyde in the presence of organic carboxylic acid salts of Group IB metals.

A Model of Loudness Applicable to Time-Varying Sounds

Abstract: Previously we described a model for calculating the loudness of steady sounds from their spectrum. Here a new version of the model is presented, which uses a waveform as its input. The stages of the model are as follows. (a) A finite impulse response filter representing transfer through the outer and middle ear. (b) Calculation of the short-term spectrum using the fast Fourier transform (FFT). To give adequate spectral resolution at low frequencies, combined with adequate temporal resolution at high frequencies, six FFTs are calculated in parallel, using longer signal segments for low frequencies and shorter segments for higher frequencies. (c) Calculation of an excitation pattern from the physical spectrum. (d) Transformation of the excitation pattern to a specific loudness pattern. (e) Determination of the area under the specific loudness pattern. This gives a value for the instantaneous loudness. The short-term perceived loudness is calculated from the instantaneous loudness using an averaging mechanism similar to an automatic gain control system, with attack and release times. Finally the overall loudness impression is calculated from the short-term loudness using a similar averaging mechanism, but with longer attack and release times. The new model gives very similar predictions to our earlier model for steady sounds. In addition, it can predict the loudness of brief sounds as a function of duration and the overall loudness of sounds that are amplitude modulated at various rates.

Swipe: a sawtooth waveform inspired pitch estimator for speech and music

Abstract: A Sawtooth Waveform Inspired Pitch Estimator (SWIPE) has been developed for processing speech and music. SWIPE is shown to outperform existing algorithms on several publicly available speech/musical-instruments databases and a disordered speech database. SWIPE estimates the pitch as the fundamental frequency of the sawtooth waveform whose spectrum best matches the spectrum of the input signal. A decaying cosine kernel provides an extension to older frequency-based, sieve-type estimation algorithms by providing smooth peaks with decaying amplitudes to correlate with the harmonics of the signal. An improvement on the algorithm is achieved by using only the first and prime harmonics, which significantly reduces subharmonic errors commonly found in other pitch estimation algorithms.

Monaural speech segregation based on pitch tracking and

Abstract: inability to deal with signals in the high -freq uency range. Psychoacoustic evidence sugge sts that different perceptual mechanisms are involved to handle resolved and unresolved hannonics. We propose a system for speech segre gation that deals with low-frequency and high-frequency signals differently. For low-frequency signals , our model generates segm ents based on temporal continuity and cross-channel correlation , and groups them according to periodicity. For high-frequency signals, the model generates segments based on common amplitude modulation (AM) in addition to temporal contin uity, and grou ps them according to AM repetition rates. Underlying the group ing process is a p itch contour that is first est imated from segregated speech based on global pitch and then verified by psychoacoustic constraints. Our system is systematically evaluated, and it yields substantially better performan ce than previous CASA systems, especially in the high-frequency range.

Monaural speech segregation based on pitch tracking and amplitude modulation

Abstract: Segregating speech from one monaural recording has proven to be very challenging. Monaural segregation of voiced speech has been studied in previous systems that incorporate auditory scene analysis principles. A major problem for these systems is their inability to deal with the high-frequency part of speech. Psychoacoustic evidence suggests that different perceptual mechanisms are involved in handling resolved and unresolved harmonics. We propose a novel system for voiced speech segregation that segregates resolved and unresolved harmonics differently. For resolved harmonics, the system generates segments based on temporal continuity and cross-channel correlation, and groups them according to their periodicities. For unresolved harmonics, it generates segments based on common amplitude modulation (AM) in addition to temporal continuity and groups them according to AM rates. Underlying the segregation process is a pitch contour that is first estimated from speech segregated according to dominant pitch and then adjusted according to psychoacoustic constraints. Our system is systematically evaluated and compared with pervious systems, and it yields substantially better performance, especially for the high-frequency part of speech.