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PubMed2003

Binaural sensitivity as a function of interaural electrode position with a bilateral cochlear implant user.

Experiments were conducted with a single, bilateral cochlear implant user to examine interaural level and time-delay cues that putatively underlie the design and efficacy of bilateral implant systems. The subject's two implants were of different types but custom equipment allowed presentation of controlled bilateral stimuli, particularly those with specified interaural time difference (ITD) and interaural level difference (ILD) cues. A lateralization task was used to measure the effect of these cues on the perceived location of the sensations elicited. For trains of fixed-amplitude, biphasic current pulses at 100 pps, the subject demonstrated sensitivity to an ITD of 300 micros, providing evidence of access to binaural information. The choice of bilateral electrode pair greatly influenced ITD sensitivity, suggesting that electrode pairings are likely to be an important consideration in the effort to provide binaural advantages. The selection of bilateral electrode pairs showing sensitivity to ITD was partially aided by comparisons of the pitch elicited by individual electrodes in each ear (when stimulated alone with fixed-amplitude current pulses at 813 pps): specifically, interaural electrodes with similar pitches were more likely (but not certain) to show ITD sensitivity. Significant changes in lateral position occurred with specific electrode pairs. With five bilateral electrode pairs of 14 tested, ITDs of 300 and 600 micros moved an auditory image significantly from right to left. With these same pairs, ILD changes of approximately 11% of the dynamic range (in microApp) moved an auditory image from the far left to the far right-significantly farther than the nine pairs not showing significant ITD sensitivity. However, even these nine pairs did show response changes as a function of the interaural (or confounding monaural) level cue. Overall, insofar as the access to bilateral cues demonstrated herein generalizes to other subjects, it provides hope that the normal binaural advantages for speech recognition and sound localization can be made available to bilateral implant users.

Acoustic Stimulation↗
PubMed2003

Time course of loudness recalibration: implications for loudness enhancement.

Loudness recalibration, the effect of a relatively loud 2500-Hz recalibrating tone on the loudness of a relatively soft 2500-Hz target tone, was measured as a function of the interstimulus interval (ISI) between them. The loudness of the target tone, assessed by a 500-Hz comparison tone, declined when the ISI equaled or exceeded about 200 ms and leveled off at an ISI of about 700 ms. Notably, the target tone's loudness did not change significantly at very short ISIs (< 150 ms). The latter result is incompatible with the literature reporting loudness enhancement in this time window, but is compatible with the suggestion made by Scharf, Buus, and Nieder [J. Acoust. Soc. Am. 112, 807-810 (2002)] that early measurements of enhancement were contaminated by the influence of the recalibrating tone on the comparison tone when the two shared the same frequency. In a second experiment the frequency of the comparison tone was changed to 2500 Hz and the results of a loudness enhancement paradigm was successfully predicted from the time course of recalibration obtained in experiment 1.

Acoustic Stimulation↗
PubMed2003

Cochlear compression estimates from measurements of distortion-product otoacoustic emissions.

Evidence of the compressive growth of basilar-membrane displacement can be seen in distortion-product otoacoustic emission (DPOAE) levels measured as a function of stimulus level. When the levels of the two stimulus tones (f1 and f2) are related by the formula L1 = 39 dB + 0.4 x L2 [Kummer et al., J. Acoust. Soc. Am. 103, 3431-3444 (1998)] the shape of the function relating DPOAE level to L2 is similar (up to an L2 of 70 dB SPL) to the classic Fletcher and Munson [J. Acoust. Soc. Am. 9, 1-10 (1933)] loudness function when plotted on a logarithmic scale. Explicit estimates of compression have been derived based on recent DPOAE measurements from the laboratory. If DPOAE growth rate is defined as the slope of the DPOAE I/O function (in dB/dB), then a cogent definition of compression is the reciprocal of the growth rate. In humans with normal hearing, compression varies from about 1 at threshold to about 4 at 70 dB SPL. With hearing loss, compression is still about 1 at threshold, but grows more slowly above threshold. Median DPOAE I/O data from ears with normal hearing, mild loss, and moderate loss are each well fit by log functions. When the I/O function is logarithmic, then the corresponding compression is a linear function of stimulus level. Evidence of cochlear compression also exists in DPOAE suppression tuning curves, which indicate the level of a third stimulus tone (f3) that reduces DPOAE level by 3 dB. All three stimulus tones generate compressive growth within the cochlea; however, only the relative compression (RC) of the primary and suppressor responses is observable in DPOAE suppression data. An RC value of 1 indicates that the cochlear responses to the primary and suppressor components grow at the same rate. In normal ears, RC rises to 4, when f3 is an octave below f2. The similarities between DPOAE and loudness compression estimates suggest the possibility of predicting loudness growth from DPOAEs; however, intersubject variability makes such predictions difficult at this time.

Auditory Threshold↗
PubMed2003

Spectral pattern, harmonic relations, and the perceptual grouping of low-numbered components.

Mistuning a harmonic increases its salience and produces an exaggerated change in its pitch. The effects on component grouping of spectral pattern, global pitch, and local harmonicity were explored using these phenomena. Stimuli were either harmonic (F0 = 200 Hz) or frequency shifted by 25% of F0. Component 1 or 2 was replaced by one of a set of sinusoidal probes in the same spectral region. Listeners either matched the probe pitch by adjusting the frequency of a pure tone (experiments 1 and 3) or matched the probe loudness by adjusting the level of a tone of identical frequency (experiment 2). Probe positions corresponding to greatest perceptual fusion were estimated from the variations in pitch shift and loudness across frequency. Both measures gave similar estimates. For harmonic stimuli, fusion was greatest at harmonic values. For shifted stimuli, fusion was greatest close to the suboctave (225 Hz) and the frequency (450 Hz) of component 2. The latter value moved downwards to near 433 Hz (2:3 ratio with component 3) when component 1 was removed. Together, these results indicate that the lowest component of a shifted complex is grouped by local harmonicity, whereas the higher components are grouped by common spectral spacing. Global pitch did not influence component grouping.

Cochlear Nerve↗
PubMed2003

Phase effects in masking: within- versus across-channel processes.

The effects of bandwidth and component phase on masking were investigated using 200-ms narrowband (1-ERB(N)) and broadband (5-ERB(N)) cosine-phase (CP) and random-phase (RP) harmonic complex maskers, centered at 1 or 6 kHz. A continuous notched-noise was used to restrict off-frequency listening. The masker fundamental frequency (F0) was 25 Hz. In experiment 1, thresholds were measured for sinusoidal signals at 1 and 6 kHz, gated with the maskers. Thresholds were lower in the CP than in the RP masker, for both bandwidths, but the effect was markedly greater for the wider bandwidth. For the CP maskers, thresholds were markedly lower for the 5-ERB(N) than for the 1-ERB(N) bandwidth; for the RP maskers, there was a small effect in the opposite direction. Experiment 2 used 1- and 6-kHz CP maskers. The masker components in the ERB(N) around the signal frequency were presented to one ear, and the remaining components were presented contralaterally. Thresholds were much higher than when all components were presented to the same ear, and were higher than for the 1-ERB(N) masker alone, suggesting that the low thresholds for broadband monaural presentation do not depend on "high level" across-channel comparisons. Simultaneous masked thresholds could be predicted well using a model based on a simulated auditory filter, a level-dependent compressive nonlinearity, and a sliding temporal integrator; it was not necessary to assume the involvement of across-channel processes or of selective listening in the masker dips.

Adult↗
PubMed2003

A phenomenological model for the responses of auditory-nerve fibers. II. Nonlinear tuning with a frequency glide.

A computational model was developed to simulate the responses of auditory-nerve (AN) fibers in cat. The model's signal path consisted of a time-varying bandpass filter; the bandwidth and gain of the signal path were controlled by a nonlinear feed-forward control path. This model produced realistic response features to several stimuli, including pure tones, two-tone combinations, wideband noise, and clicks. Instantaneous frequency glides in the reverse-correlation (revcor) function of the model's response to broadband noise were achieved by carefully restricting the locations of the poles and zeros of the bandpass filter. The pole locations were continuously varied as a function of time by the control signal to change the gain and bandwidth of the signal path, but the instantaneous frequency profile in the revcor function was independent of sound pressure level, consistent with physiological data. In addition, this model has other important properties, such as nonlinear compression, two-tone suppression, and reasonable Q10 values for tuning curves. The incorporation of both the level-independent frequency glide and the level-dependent compressive nonlinearity into a phenomenological model for the AN was the primary focus of this work. The ability of this model to process arbitrary sound inputs makes it a useful tool for studying peripheral auditory processing.

Auditory Pathways↗
PubMed2003

Perceptual segregation of competing speech sounds: the role of spatial location.

Culling and Summerfield [J. Acoust Soc. Am. 92, 785-797 (1995)] showed that listeners could not use ongoing interaural time differences (ITDs) to achieve source segregation. The present experiments tested a free-field analog of their experiment. The stimuli consisted of narrow bands of noise, pairs of which represented the first and second formants of the whispered vowels "ar," "ee," "er," and "oo." A target noise-band pair (vowel) was presented at various angles on the listeners' left while a complementary distracter was presented on the listeners' right. Listeners correctly identified the target vowel in the free-field well above chance. Performance remained well above chance in headphone experiments that retained spatial cues but eliminated reverberations and head movements. The full range of cues that normally determine perceived spatial location provided sufficient information for segregation. Further experiments, which systematically evaluated the contribution of these cues in isolation and in combination, showed that some listeners, following training, exhibited the ability to segregate based on ongoing ITDs alone. Substantial individual differences were observed. The results show that listeners can use spatial cues to segregate simultaneous sound sources.

Adolescent↗
PubMed2003

A measure of internal noise based on sample discrimination.

Internal noise is often inferred from the difference between observed performance and optimum performance in detection and discrimination tasks. It can be measured directly in some cases by observing the extent to which a change in external variability impacts performance. In the studies reported here, external variability was added to an intensity discrimination task by adding a Gaussian random variable with zero mean to the overall level presented in each interval of a two-interval forced-choice task. The standard deviation of the random variable was set to half the mean difference between the levels in the two intervals, resulting in d'(ideal) = 2. As the mean difference and the corresponding standard deviation of the random variable decreased in size, performance was increasingly limited by internal noise, permitting a reliable estimate of internal noise to be obtained. This can be viewed as a sample discrimination task, with one component per sample. In the first study, performance was measured using 2-kHz tones presented at an average level of 70 dB SPL, with mean differences between distributions ranging from 0.1 to 2.2 dB in steps of 0.3 dB. The distributions were either Gaussian in level or in power. Conditions with no external variability were used to obtain a psychometric function. In the second study, performance was measured using 2-kHz tones presented at average levels of 50 and 90 dB SPL, with mean differences ranging from 0.4 to 2.2 dB in steps of 0.6 dB. In both studies, the measure of internal noise was highly reliable and in good agreement with the intensity difference limen (DL) estimated from the psychometric function. Analyses suggest that this measure could be used to estimate the mean difference between the decision distributions as well as the amount of internal noise in cases where the mean difference between the distributions is unknown.

Adolescent↗
PubMed2003

Application of loudness models to sound processing for cochlear implants.

A new paradigm for processing sound signals for multiple-electrode cochlear implants is introduced, and results are presented from an initial psychophysical evaluation of its effect on the perceived loudness of complex sounds. A real-time processing scheme based on this paradigm, called SpeL, has been developed primarily to improve control of loudness for implant users. SpeL differs from previous schemes in several ways. Most importantly, it incorporates a published numerical model which predicts the loudness perceived by implant users for complex patterns of pulsatile electric stimulation as a function of the pulses' physical parameters. This model is controlled by the output of a corresponding model that estimates the loudness perceived by normally hearing listeners for complex sounds. The latter model produces an estimate of the specific loudness arising from an acoustic signal. In SpeL, the specific loudness function, which describes the contribution to total loudness of each of a number of frequency bands (or cochlear positions), is converted to a pattern of electric stimulation on an appropriate set of electrodes. By application of the loudness model for electric stimulation, this pattern is designed to produce a specific loudness function for the implant user which approximates that produced by the normal-hearing model for the same input signal. The results of loudness magnitude estimation experiments with five users of the SpeL scheme confirmed that the psychophysical functions relating overall loudness perceived to input sound level for five complex acoustic signals were, on average, very similar to those for normal hearing.

Aged↗
PubMed2003

Modulation masking produced by complex tone modulators.

Thresholds were measured for detecting sinusoidal amplitude modulation in the presence of a complex-tone masker modulation. Both modulations were applied to the same sinusoidal carrier. Two different masker modulations were used: (i) a pair of components beating at the difference frequency and (ii) a three-tone complex producing a sinusoidal amplitude modulation of the modulation depth at the difference frequency between adjacent components. Both maskers show a periodicity in the waveform that is not contained in the envelope spectrum itself but can be observed when the envelope of the envelope, referred to as the "venelope" [Ewert et al., J. Acoust. Soc. Am. 112, 2921-2931 (2002)], is calculated. For a signal frequency equal to the masker-venelope periodicity, modulation depth at threshold was measured as a function of the signal phase relative to the phase of the masker-venelope component. Signal frequencies of 5, 30, and 90 Hz were used. It was found that masking was phase dependent for all three signal frequencies. Thresholds were lower for the in-phase condition, where maxima in the signal waveform coincided with maxima in the masker-venelope waveform, than for the antiphase condition. The maximum threshold difference was 15 dB. The results are in contrast to recent data [Moore et al., J. Acoust. Soc. Am. 106, 908-918 (1999)], where lowest thresholds were reported for the antiphase condition in a similar experiment. The present data are in line with the idea that a nonlinearity prior to a modulation filterbank extracts the venelope of the masker modulator. However, a compressive nonlinearity such as that associated with the processing on the basilar membrane cannot account for the empirical findings, since it predicts the opposite phase effect.

Adult↗
PubMed2003

Desynchronization of electrically evoked auditory-nerve activity by high-frequency pulse trains of long duration.

Rubinstein et al. [Hear. Res. 127, 108-118 (1999)] suggested that the neural representation of the waveforms of electric stimuli might be improved by introducing an ongoing, high-rate, desynchronizing pulse train (DPT). A DPT may desynchronize neural responses to electric stimulation in a manner similar to spontaneous activity in a healthy ear. To test this hypothesis, responses of auditory-nerve fibers (ANFs) to 10-min-long electric pulse trains (5 kpps) were recorded from acutely deafened, anesthetized cats. Stimuli were delivered via an intracochlear electrode, and their amplitude was chosen to elicit a response in most ANFs. Responses to pulse trains showed pronounced adaptation during the first 1-2 min, followed by either a sustained response or cessation of spike discharges for the remainder of the stimulus. The adapted discharge rates showed a broad distribution across the ANF population like spontaneous activity. However, a higher proportion of fibers (46%) responded to the DPT at rates below 5 spikes/s than for spontaneous activity, and 12% of the fibers responded at higher rates than any spontaneously active fiber. Interspike interval histograms of sustained responses for some fibers had Poisson-like (exponential) shapes, resembling spontaneous activity, while others exhibited preferred intervals and, occasionally, bursting. Simultaneous recordings from pairs of fibers revealed no evidence of correlated activity, suggesting that the DPT does desynchronize the auditory nerve activity. Overall, these results suggest that responses to an ongoing DPT resemble spontaneous activity in a normal ear for a substantial fraction of the ANFs.

Acoustic Stimulation↗
PubMed2003

Improved temporal coding of sinusoids in electric stimulation of the auditory nerve using desynchronizing pulse trains.

Rubinstein et al. [Hearing Res. 127, 108-118 (1999)] suggested that the representation of electric stimulus waveforms in the temporal discharge patterns of auditory-nerve fiber (ANF) might be improved by introducing an ongoing, high-rate, desynchronizing pulse train (DPT). To test this hypothesis, activity of ANFs was studied in acutely deafened, anesthetized cats in response to 10-min-long, 5-kpps electric pulse trains that were sinusoidally modulated for 400 ms every second. Two classes of responses to sinusoidal modulations of the DPT were observed. Fibers that only responded transiently to the unmodulated DPT showed hyper synchronization and narrow dynamic ranges to sinusoidal modulators, much as responses to electric sinusoids presented without a DPT. In contrast, fibers that exhibited sustained responses to the DPT were sensitive to modulation depths as low as 0.25% for a modulation frequency of 417 Hz. Over a 20-dB range of modulation depths, responses of these fibers resembled responses to tones in a healthy ear in both discharge rate and synchronization index. This range is much wider than the dynamic range typically found with electrical stimulation without a DPT, and comparable to the dynamic range for acoustic stimulation. These results suggest that a stimulation strategy that uses small signals superimposed upon a large DPT to encode sounds may evoke temporal discharge patterns in some ANFs that resemble responses to sound in a healthy ear.

Animals↗
PubMed2003

Interactions between test- and inducer-tone durations in induced loudness reduction.

A tone usually declines in loudness when preceded by a more intense inducer tone. This phenomenon is called "loudness recalibration" or "induced loudness reduction" (ILR). The present study investigates how ILR depends on level, loudness, and duration. A 2AFC procedure was used to obtain loudness matches between 2500-Hz comparison tones and 500-Hz test tones at 60 and 70 dB SPL, presented with and without preceding 500-Hz inducer tones. For 200-ms test and comparison tones, the amount of ILR did not depend on inducer level (set at 80 dB SPL and above), but ILR was greater with 200- than with 5-ms inducers, even when both were equally loud. For 5-ms tones, ILR was as great with 5- as with 200-ms inducers and about as great as when test and inducer tones both lasted 200 ms. These results suggest that (1) neither the loudness nor the SPL of the inducer alone governs ILR, and (2) inducer duration must equal or exceed test-tone duration to yield maximal amounts of ILR. Further analysis indicates that the efferent system may be partly responsible for ILR of 200-ms test tones, but is unlikely to account for ILR of 5-ms tones.

Adult↗
PubMed2003

High-rate conditioning pulse trains in cochlear implants: dynamic range measures with sinusoidal stimuli.

The addition of a continuous, unmodulated, high-rate pulse train to the electrical signals of cochlear implant recipients results in statistically significant increases in psychophysical dynamic range (41 out of 46 electrode pairs tested). The observed increases in dynamic range are thought to result from nerve conditioning by appropriate levels of high-rate pulse train. Five dynamic range profiles are characterized, defining the different responses of dynamic range observed with increasing levels of the conditioner. Four of the five profiles demonstrate increases in dynamic range, with three showing behavior consistent with stochastic resonance. One profile depicts evidence of adaptation in response to higher levels of the conditioner, with a recovery period lasting throughout the duration (on the scale of tens of minutes) of experimentation. Dynamic range profiles are shown to be similar across sinusoidal frequencies (202, 515, and 1031 Hz) but potentially different across electrode pairs (electrodes 1-2, 7-8, and 15-16). Correlation analysis does not reveal any predictors of optimal conditioner level or amount of dynamic range increase with the conditioner.

Adult↗
PubMed2003

Tone decay for hearing-impaired listeners with and without dead regions in the cochlea.

For people with normal hearing, a sustained tone with a frequency within the standard audiometric range remains audible when presented at a level well above threshold. However, for a pure tone with frequency close to the upper limit of hearing (well above 8 kHz), the loudness may decrease within seconds and the tone may decay to inaudibility, even when presented at a level between 20 and 40 dB SL. Scharf [in Hearing Research and Theory, edited by J. V. Tobias and E. D. Schubert (Academic, New York, 1983), Vol. 2, pp. 1-53] suggested that marked loudness adaptation only occurs when the excitation pattern evoked by a tone is spatially limited. The upper limit of hearing may be comparable to the boundary of a "dead region," which is a region with a complete loss of inner hair cell (IHC) and/or neural function. The present study investigated the perceived decay of pure tones for 9 normal-hearing subjects and 12 subjects with moderate to severe sensorineural hearing loss, using a wide range of frequencies (0.125-12 kHz). A dead region was diagnosed for 8 of the 12 subjects. No consistent association was found between the degree of tone decay and the presence of a dead region. Subjects with dead regions did not experience significantly more tone decay than subjects with comparable absolute thresholds but without a dead region, even when the frequency of the tone fell within or close to the edge of a dead region. For severely hearing-impaired subjects, spatial restriction of the excitation pattern was neither necessary nor sufficient to lead to tone decay. The prevalence of tone decay was not well predicted by the audiometric threshold at the test frequency. It is proposed that tone decay depends on the physiological condition of the place in the cochlea where the tone is detected, which, in a case involving a dead region, is the place adjacent to the dead region. The prevalence of tone decay increased when the audiometric threshold was above 50 dB HL in the frequency region where the tone was detected.

Adult↗
PubMed2004

Performance of time- and frequency-domain binaural beamformers based on recorded signals from real rooms.

Extraction of a target sound source amidst multiple interfering sound sources is difficult when there are fewer sensors than sources, as is the case for human listeners in the classic cocktail-party situation. This study compares the signal extraction performance of five algorithms using recordings of speech sources made with three different two-microphone arrays in three rooms of varying reverberation time. Test signals, consisting of two to five speech sources, were constructed for each room and array. The signals were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio of the output. A frequency-domain minimum-variance distortionless-response beamformer outperformed the time-domain based Frost beamformer and generalized sidelobe canceler for all tests with two or more interfering sound sources, and performed comparably or better than the time-domain algorithms for tests with one interfering sound source. The frequency-domain minimum-variance algorithm offered performance comparable to that of the Peissig-Kollmeier binaural frequency-domain algorithm, but with much less distortion of the target signal. Comparisons were also made to a simple beamformer. In addition, computer simulations illustrate that, when processing speech signals, the chosen implementation of the frequency-domain minimum-variance technique adapts more quickly and accurately than time-domain techniques.

Algorithms↗
PubMed2004

Development of a new standard laboratory protocol for estimation of the field attenuation of hearing protection devices: sample size necessary to provide acceptable reproducibility.

The mandate of ASA Working Group S12/WG11 has been to develop "laboratory and/or field procedure(s) that yield useful estimates of field performance" of hearing protection devices (HPDs). A real-ear attenuation at threshold procedure was selected, devised, tested for one earmuff and three earplugs via an interlaboratory study involving five laboratories and 147 subjects, and incorporated into a new standard that was approved in 1997 [Royster et al., "Development of a new standard laboratory protocol for estimating the field attenuation of hearing protection devices. Part I. Research of Working Group 11, Accredited Standards Committee S 12, Noise," J. Acoust. Soc. Am. 99, 1506-1526; ANSI, S12.6-1997, "American National Standard method for measuring real-ear attenuation of hearing protectors" (American National Standards Institute, New York, 1997)]. The subject-fit methodology of ANSI S12.6-1997 relies upon listeners who are audiometrically proficient, but inexperienced in the use of HPDs. Whenever a new method is adopted, it is important to know the effects of variability on the power of the measurements. In evaluation of protector noise reduction determined by experimenter-fit, informed-user-fit, and subject-fit methods, interlaboratory reproducibility was found to be best for the subject-fit method. Formulas were derived for determining the minimum detectable difference between attenuation measurements and for determining the number of subjects necessary to achieve a selected level of precision. For a precision of 6 dB, the study found that the minimum number of subjects was 4 for the Bilsom UF-1 earmuff, 10 for the E.A.R Classic earplug, 31 for the Willson EP100 earplug, and 22 for the PlasMed V-51R earplug.

Auditory Threshold↗