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To examine difficulties experienced by cochlear implant (CI) users when perceiving non-native speech, intelligibility of non-native speech was compared in conditions with single and multiple alternating talkers. Compared to listeners with normal hearing, no rapid talker-dependent adaptation was observed and performance was approximately 40% lower for CI users following increased exposure in both talker conditions. Results suggest that lower performance for CI users may stem from combined effects of limited spectral resolution, which diminishes perceptible differences across accents, and limited access to talker-specific acoustic features of speech, which reduces the ability to adapt to non-native speech in a talker-dependent manner. [ABSTRACT FROM AUTHOR]

Long-term loudness perception of a sound has been presumed to depend on the spatial distribution of activated auditory nerve fibers as well as their temporal firing pattern. The relative contributions of those two factors were investigated by measuring loudness adaptation to sinusoidally amplitude-modulated 12-kHz tones. The tones had a total duration of 180 s and were either unmodulated or 100%-modulated at one of three frequencies (4, 20, or 100 Hz), and additionally varied in modulation depth from 0% to 100% at the 4-Hz frequency only. Every 30 s, normal-hearing subjects estimated the loudness of one of the stimuli played at 15 dB above threshold in random order. Without any amplitude modulation, the loudness of the unmodulated tone after 180 s was only 20% of the loudness at the onset of the stimulus. Amplitude modulation systematically reduced the amount of loudness adaptation, with the 100%-modulated stimuli, regardless of modulation frequency, maintaining on average 55%-80% of the loudness at onset after 180 s. Because the present low-frequency amplitude modulation produced minimal changes in long-term spectral cues affecting the spatial distribution of excitation produced by a 12-kHz pure tone, the present result indicates that neural synchronization is critical to maintaining loudness perception over time. [ABSTRACT FROM AUTHOR]

Understanding speech-in-noise is difficult for most cochlear implant (CI) users. Speech-in-noise segregation cues are well understood for acoustic hearing but not for electric hearing. This study investigated the effects of stimulation rate and onset delay on synthetic vowel-in-noise recognition in CI subjects. In experiment I, synthetic vowels were presented at 50, 145, or 795 pulse/s and noise at the same three rates, yielding nine combinations. Recognition improved significantly if the noise had a lower rate than the vowel, suggesting that listeners can use temporal gaps in the noise to detect a synthetic vowel. This hypothesis is supported by accurate prediction of synthetic vowel recognition using a temporal integration window model. Using lower rates a similar trend was observed in normal hearing subjects. Experiment II found that for CI subjects, a vowel onset delay improved performance if the noise had a lower or higher rate than the synthetic vowel. These results show that differing rates or onset times can improve synthetic vowel-in-noise recognition, indicating a need to develop speech processing strategies that encode or emphasize these cues. [ABSTRACT FROM AUTHOR]

Across bilateral cochlear implants, contralateral threshold shift has been investigated as a function of electrode difference between the masking and probe electrodes. For contralateral electric masking, maximum threshold elevations occurred when the position of the masker and probe electrode was approximately place-matched across ears. The amount of masking diminished with increasing masker-probe electrode separation. Place-dependent masking occurred in both sequentially implanted ears, and was not affected by the masker intensity or the time delay from the masker onset. When compared to previous contralateral masking results in normal hearing, the similarities between place-dependent central masking patterns suggest comparable mechanisms of overlapping excitation in the central auditory nervous system. [ABSTRACT FROM AUTHOR]

The addition of low-passed (LP) speech or even a tone following the fundamental frequency (F0) of speech has been shown to benefit speech recognition for cochlear implant (CI) users with residual acoustic hearing. The mechanisms underlying this benefit are still unclear. In this study, eight bimodal subjects (CI users with acoustic hearing in the non-implanted ear) and eight simulated bimodal subjects (using vocoded and LP speech) were tested on vowel and consonant recognition to determine the relative contributions of acoustic and phonetic cues, including F0, to the bimodal benefit. Several listening conditions were tested (CI/Vocoder, LP, TF0-env, CI/Vocoder + LP, CI/Vocoder + TF0-env). Compared with CI/Vocoder performance, LP significantly enhanced both consonant and vowel perception, whereas a tone following the F0 contour of target speech and modulated with an amplitude envelope of the maximum frequency of the F0 contour (TF0-env) enhanced only consonant perception. Information transfer analysis revealed a dual mechanism in the bimodal benefit: The tone representing F0 provided voicing and manner information, whereas LP provided additional manner, place, and vowel formant information. The data in actual bimodal subjects also showed that the degree of the bimodal benefit depended on the cutoff and slope of residual acoustic hearing. [ABSTRACT FROM AUTHOR]

The present study examined the effect of combined spectral and temporal enhancement on speech recognition by cochlear-implant (CI) users in quiet and in noise. The spectral enhancement was achieved by expanding the short-term Fourier amplitudes in the input signal. Additionally, a variation of the Transient Emphasis Spectral Maxima (TESM) strategy was applied to enhance the short-duration consonant cues that are otherwise suppressed when processed with spectral expansion. Nine CI users were tested on phoneme recognition tasks and ten CI users were tested on sentence recognition tasks both in quiet and in steady, speech-spectrum-shaped noise. Vowel and consonant recognition in noise were significantly improved with spectral expansion combined with TESM. Sentence recognition improved with both spectral expansion and spectral expansion combined with TESM. The amount of improvement varied with individual CI users. Overall the present results suggest that customized processing is needed to optimize performance according to not only individual users but also listening conditions. [ABSTRACT FROM AUTHOR]

Cochlear implant (CI) users have been shown to benefit from residual low-frequency hearing, specifically in pitch related tasks. It remains unclear whether this benefit is dependent on fundamental frequency (F0) or other acoustic cues. Three experiments were conducted to determine the role of F0, as well as its frequency modulated (FM) and amplitude modulated (AM) components, in speech recognition with a competing voice. In simulated CI listeners, the signal-to-noise ratio was varied to estimate the 50% correct response. Simulation results showed that the F0 cue contributes to a significant proportion of the benefit seen with combined acoustic and electric hearing, and additionally that this benefit is due to the FM rather than the AM component. In actual CI users, sentence recognition scores were collected with either the full F0 cue containing both the FM and AM components or the 500-Hz low-pass speech cue containing the F0 and additional harmonics. The F0 cue provided a benefit similar to the low-pass cue for speech in noise, but not in quiet. Poorer CI users benefited more from the F0 cue than better users. These findings suggest that F0 is critical to improving speech perception in noise in combined acoustic and electric hearing. [ABSTRACT FROM AUTHOR]

Residual acoustic hearing can be preserved in the same ear following cochlear implantation with minimally traumatic surgical techniques and short-electrode arrays. The combined electric-acoustic stimulation significantly improves cochlear implant performance, particularly speech recognition in noise. The present study measures simultaneous masking by electric pulses on acoustic pure tones, or vice versa, to investigate electric-acoustic interactions and their underlying psychophysical mechanisms. Six subjects, with acoustic hearing preserved at low frequencies in their implanted ear, participated in the study. One subject had a fully inserted 24 mm Nucleus Freedom array and five subjects had Iowa/Nucleus hybrid implants that were only 10 mm in length. Electric masking data of the long-electrode subject showed that stimulation from the most apical electrodes produced threshold elevations over 10 dB for 500, 625, and 750 Hz probe tones, but no elevation for 125 and 250 Hz tones. On the contrary, electric stimulation did not produce any electric masking in the short-electrode subjects. In the acoustic masking experiment, 125-750 Hz pure tones were used to acoustically mask electric stimulation. The acoustic masking results showed that, independent of pure tone frequency, both long- and short-electrode subjects showed threshold elevations at apical and basal electrodes. The present results can be interpreted in terms of underlying physiological mechanisms related to either place-dependent peripheral masking or place-independent central masking. [ABSTRACT FROM AUTHOR]

Bilateral cochlear implant (BiCI) users gain an advantage in noisy situations from a second implant, but their bilateral performance falls short of normal hearing listeners. Channel interactions due to overlapping electrical fields between electrodes can impair speech perception, but its role in limiting binaural hearing performance has not been well characterized. To address the issue, binaural masking level differences (BMLD) for a 125 Hz tone in narrowband noise were measured using a pair of pitch-matched electrodes while simultaneously presenting the same masking noise to adjacent electrodes, representing a more realistic stimulation condition compared to prior studies that used only a single electrode pair. For five subjects, BMLDs averaged 8.9 ± 1.0 dB (mean ± s.e.) in single electrode pairs but dropped to 2.1 ± 0.4 dB when presenting noise on adjacent masking electrodes, demonstrating a negative impact of the additional maskers. Removing the masking noise from only the pitch-matched electrode pair not only lowered thresholds but also resulted in smaller BMLDs. The degree of channel interaction estimated from auditory nerve evoked potentials in three subjects was significantly and negatively correlated with BMLD. The data suggest that if the amount of channel interactions can be reduced, BiCI users may experience some performance improvements related to binaural hearing. [ABSTRACT FROM AUTHOR]

At present commercially available bilateral cochlear implants (CIs) improve their users’ speech understanding in noise but they employ two independent speech processors that cannot provide accurate and appropriate interaural level and time differences as seen binaurally in normal hearing (NH) listeners. Previous work suggests that binaural cues are accessible to bilateral CI users when presented to single pairs of pitch-matched electrodes, but the scope was limited and the mechanisms remained unclear. In this study, binaural masking level differences (BMLDs) were measured in five bilateral Nucleus-24 CI users over multiple pairs of pitch-matched electrodes. Average BMLD was 4.6±4.9 dB, but large individual variability prevented significance (p=0.09). Considering just the 125 Hz condition, as in previous work, phase (N0S0 vs N0Sπ) and electrode effects were significant. Compared with simulated bilateral CI users, actual bilateral CI users had proportionally higher thresholds for N0Sπ than N0S0. Together the present results suggest that the performance gap in BMLDs between CI and NH listeners is not due to a lack of sufficient acoustic cues in the temporal envelope domain but to a true binaural deficit related to a central mechanism in deprived binaural processing. [ABSTRACT FROM AUTHOR]

The possibility that “dead regions” or “spectral holes” can account for some differences in performance between bilateral cochlear implant (CI) users and normal-hearing listeners was explored. Using a 20-band noise-excited vocoder to simulate CI processing, this study examined effects of spectral holes on speech reception thresholds (SRTs) and spatial release from masking (SRM) in difficult listening conditions. Prior to processing, stimuli were convolved through head-related transfer-functions to provide listeners with free-field directional cues. Processed stimuli were presented over headphones under binaural or monaural (right ear) conditions. Using Greenwood’s [(1990). J. Acoust. Soc. Am. 87, 2592–2605] frequency-position function and assuming a cochlear length of 35 mm, spectral holes were created for variable sizes (6 and 10 mm) and locations (base, middle, and apex). Results show that middle-frequency spectral holes were the most disruptive to SRTs, whereas high-frequency spectral holes were the most disruptive to SRM. Spectral holes generally reduced binaural advantages in difficult listening conditions. These results suggest the importance of measuring dead regions in CI users. It is possible that customized programming for bilateral CI processors based on knowledge about dead regions can enhance performance in adverse listening situations. [ABSTRACT FROM AUTHOR]

A cochlear implant vocoder was used to evaluate relative contributions of spectral and binaural temporal fine-structure cues to speech intelligibility. In Study I, stimuli were vocoded, and then convolved through head related transfer functions (HRTFs) to remove speech temporal fine structure but preserve the binaural temporal fine-structure cues. In Study II, the order of processing was reversed to remove both speech and binaural temporal fine-structure cues. Speech reception thresholds (SRTs) were measured adaptively in quiet, and with interfering speech, for unprocessed and vocoded speech (16, 8, and 4 frequency bands), under binaural or monaural (right-ear) conditions. Under binaural conditions, as the number of bands decreased, SRTs increased. With decreasing number of frequency bands, greater benefit from spatial separation of target and interferer was observed, especially in the 8-band condition. The present results demonstrate a strong role of the binaural cues in spectrally degraded speech, when the target and interfering speech are more likely to be confused. The nearly normal binaural benefits under present simulation conditions and the lack of order of processing effect further suggest that preservation of binaural cues is likely to improve performance in bilaterally implanted recipients. [ABSTRACT FROM AUTHOR]

Cochlear-implant users perform far below normal-hearing subjects in background noise. Speech recognition with varying numbers of competing female, male, and child talkers was evaluated in normal-hearing subjects, cochlear-implant users, and normal-hearing subjects utilizing an eight-channel sine-carrier cochlear-implant simulation. Target sentences were spoken by a male. Normal-hearing subjects obtained considerably better speech reception thresholds than cochlear-implant subjects; the largest discrepancy was 24 dB with a female masker. Evaluation of one implant subject with normal hearing in the contralateral ear suggested that this difference is not caused by age-related disparities between the subject groups. Normal-hearing subjects showed a significant advantage with fewer competing talkers, obtaining release from masking with up to three talker maskers. Cochlear-implant and simulation subjects showed little such effect, although there was a substantial difference between the implant and simulation results with talker maskers. All three groups benefited from a voice pitch difference between target and masker, with the female talker providing significantly less masking than the male. Child talkers produced more masking than expected, given their fundamental frequency, syllabic rate, and temporal modulation characteristics. Neither a simulation nor testing in steady-state noise predicts the difficulties cochlear-implant users experience in real-life noisy situations. [ABSTRACT FROM AUTHOR]

The present study evaluated auditory-visual speech perception in cochlear-implant users as well as normal-hearing and simulated-implant controls to delineate relative contributions of sensory experience and cues. Auditory-only, visual-only, or auditory-visual speech perception was examined in the context of categorical perception, in which an animated face mouthing /ba/, /da/, or /ga/ was paired with synthesized phonemes from an 11-token auditory continuum. A three-alternative, forced-choice method was used to yield percent identification scores. Normal-hearing listeners showed sharp phoneme boundaries and strong reliance on the auditory cue, whereas actual and simulated implant listeners showed much weaker categorical perception but stronger dependence on the visual cue. The implant users were able to integrate both congruent and incongruent acoustic and optical cues to derive relatively weak but significant auditory-visual integration. This auditory-visual integration was correlated with the duration of the implant experience but not the duration of deafness. Compared with the actual implant performance, acoustic simulations of the cochlear implant could predict the auditory-only performance but not the auditory-visual integration. These results suggest that both altered sensory experience and improvised acoustic cues contribute to the auditory-visual speech perception in cochlear-implant users. [ABSTRACT FROM AUTHOR]

Nonlinear sensory and neural processing mechanisms have been exploited to enhance spectral contrast for improvement of speech understanding in noise. The “companding” algorithm employs both two-tone suppression and adaptive gain mechanisms to achieve spectral enhancement. This study implemented a 50-channel companding strategy and evaluated its efficiency as a front-end noise suppression technique in cochlear implants. The key parameters were identified and evaluated to optimize the companding performance. Both normal-hearing (NH) listeners and cochlear-implant (CI) users performed phoneme and sentence recognition tests in quiet and in steady-state speech-shaped noise. Data from the NH listeners showed that for noise conditions, the implemented strategy improved vowel perception but not consonant and sentence perception. However, the CI users showed significant improvements in both phoneme and sentence perception in noise. Maximum average improvement for vowel recognition was 21.3 percentage points (p<0.05) at 0 dB signal-to-noise ratio (SNR), followed by 17.7 percentage points (p<0.05) at 5 dB SNR for sentence recognition and 12.1 percentage points (p<0.05) at 5 dB SNR for consonant recognition. While the observed results could be attributed to the enhanced spectral contrast, it is likely that the corresponding temporal changes caused by companding also played a significant role and should be addressed by future studies. [ABSTRACT FROM AUTHOR]

Speech perception in the presence of another competing voice is one of the most challenging tasks for cochlear implant users. Several studies have shown that (1) the fundamental frequency (F0) is a useful cue for segregating competing speech sounds and (2) the F0 is better represented by the temporal fine structure than by the temporal envelope. However, current cochlear implant speech processing algorithms emphasize temporal envelope information and discard the temporal fine structure. In this study, speech recognition was measured as a function of the F0 separation of the target and competing sentence in normal-hearing and cochlear implant listeners. For the normal-hearing listeners, the combined sentences were processed through either a standard implant simulation or a new algorithm which additionally extracts a slowed-down version of the temporal fine structure (called Frequency-Amplitude-Modulation-Encoding). The results showed no benefit of increasing F0 separation for the cochlear implant or simulation groups. In contrast, the new algorithm resulted in gradual improvements with increasing F0 separation, similar to that found with unprocessed sentences. These results emphasize the importance of temporal fine structure for speech perception and demonstrate a potential remedy for difficulty in the perceptual segregation of competing speech sounds. [ABSTRACT FROM AUTHOR]

This study evaluates the relative contributions of envelope and fine structure cues in both temporal and spectral domains to Mandarin tone recognition in quiet and in noise. Four sets of stimuli were created. Noise-excited vocoder speech was used to evaluate the temporal envelope. Frequency modulation was then added to evaluate the temporal fine structure. Whispered speech was used to evaluate the spectral envelope. Finally, equal-amplitude harmonics were used to evaluate the spectral fine structure. Results showed that normal-hearing listeners achieved nearly perfect tone recognition with either spectral or temporal fine structure in quiet, but only 70%–80% correct with the envelope cues. With the temporal envelope, 32 spectral bands were needed to achieve performance similar to that obtained with the original stimuli, but only four bands were necessary with the additional temporal fine structure. Envelope cues were more susceptible to noise than fine structure cues, with the envelope cues producing significantly lower performance in noise. These findings suggest that tonal pattern recognition is a robust process that can make use of both spectral and temporal cues. Unlike speech recognition, the fine structure is more important than the envelope for tone recognition in both temporal and spectral domains, particularly in noise. [ABSTRACT FROM AUTHOR]

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed. [ABSTRACT FROM AUTHOR]

Three experiments were conducted to study relative contributions of speaking rate, temporal envelope, and temporal fine structure to clear speech perception. Experiment I used uniform time scaling to match the speaking rate between clear and conversational speech. Experiment II decreased the speaking rate in conversational speech without processing artifacts by increasing silent gaps between phonetic segments. Experiment III created “auditory chimeras” by mixing the temporal envelope of clear speech with the fine structure of conversational speech, and vice versa. Speech intelligibility in normal-hearing listeners was measured over a wide range of signal-to-noise ratios to derive speech reception thresholds (SRT). The results showed that processing artifacts in uniform time scaling, particularly time compression, reduced speech intelligibility. Inserting gaps in conversational speech improved the SRT by 1.3 dB, but this improvement might be a result of increased short-term signal-to-noise ratios during level normalization. Data from auditory chimeras indicated that the temporal envelope cue contributed more to the clear speech advantage at high signal-to-noise ratios, whereas the temporal fine structure cue contributed more at low signal-to-noise ratios. Taken together, these results suggest that acoustic cues for the clear speech advantage are multiple and distributed. [ABSTRACT FROM AUTHOR]

Cochlear implants allow most patients with profound deafness to successfully communicate under optimal listening conditions. However, the amplitude modulation (AM) information provided by most implants is not sufficient for speech recognition in realistic settings where noise is typically present. This study added slowly varying frequency modulation (FM) to the existing algorithm of an implant simulation and used competing sentences to evaluate FM contributions to speech recognition in noise. Potential FM advantage was evaluated as a function of the number of spectral bands, FM depth, FM rate, and FM band distribution. Barring floor and ceiling effects, significant improvement was observed for all bands from 1 to 32 with the additional FM cue both in quiet and noise. Performance also improved with greater FM depth and rate, which might reflect resolved sidebands under the FM condition. Having FM present in low-frequency bands was more beneficial than in high-frequency bands, and only half of the bands required the presence of FM, regardless of position, to achieve performance similar to when all bands had the FM cue. These results provide insight into the relative contributions of AM and FM to speech communication and the potential advantage of incorporating FM for cochlear implant signal processing. [ABSTRACT FROM AUTHOR]

Natural spoken language processing includes not only speech recognition but also identification of the speaker’s gender, age, emotional, and social status. Our purpose in this study is to evaluate whether temporal cues are sufficient to support both speech and speaker recognition. Ten cochlear-implant and six normal-hearing subjects were presented with vowel tokens spoken by three men, three women, two boys, and two girls. In one condition, the subject was asked to recognize the vowel. In the other condition, the subject was asked to identify the speaker. Extensive training was provided for the speaker recognition task. Normal-hearing subjects achieved nearly perfect performance in both tasks. Cochlear-implant subjects achieved good performance in vowel recognition but poor performance in speaker recognition. The level of the cochlear implant performance was functionally equivalent to normal performance with eight spectral bands for vowel recognition but only to one band for speaker recognition. These results show a disassociation between speech and speaker recognition with primarily temporal cues, highlighting the limitation of current speech processing strategies in cochlear implants. Several methods, including explicit encoding of fundamental frequency and frequency modulation, are proposed to improve speaker recognition for current cochlear implant users. [ABSTRACT FROM AUTHOR]

One reason for the poor pitch performance in current cochlear-implant users may be the highly synchronized neural firing in electric hearing that lacks stochastic properties of neural firing in normal acoustic hearing. This study used three different electric stimulation patterns, jittered, probabilistic, and auditory-model-generated pulses, to mimic some aspects of the normal neural firing pattern in acoustic hearing. Pitch discrimination was measured at standard frequencies of 100, 250, 500, and 1000 Hz on three Nucleus-24 cochlear-implant users. To test the utility of the autocorrelation pitch perception model in electric hearing, one, two, and four electrodes were stimulated independently with the same patterned electric stimulation. Results showed no improvement in performance with any experimental pattern compared to the fixed-rate control. Pitch discrimination was actually worsened with the jittered pattern at low frequencies (125 and 250 Hz) than that of the control, suggesting that externally introduced stochastic properties do not improve pitch perception in electric stimulation. The multiple-electrode stimulation did not improve performance but did not degrade performance either. The present results suggest that both the right time and the right place may be needed to restore normal pitch perception in cochlear-implant users. [ABSTRACT FROM AUTHOR]

Speech recognition in noise and music perception is especially challenging for current cochlear implant users. The present study utilizes the residual acoustic hearing in the nonimplanted ear in five cochlear implant users to elucidate the role of temporal fine structure at low frequencies in auditory perception and to test the hypothesis that combined acoustic and electric hearing produces better performance than either mode alone. The first experiment measured speech recognition in the presence of competing noise. It was found that, although the residual low-frequency (<1000 Hz) acoustic hearing produced essentially no recognition for speech recognition in noise, it significantly enhanced performance when combined with the electric hearing. The second experiment measured melody recognition in the same group of subjects and found that, contrary to the speech recognition result, the low-frequency acoustic hearing produced significantly better performance than the electric hearing. It is hypothesized that listeners with combined acoustic and electric hearing might use the correlation between the salient pitch in low-frequency acoustic hearing and the weak pitch in the envelope to enhance segregation between signal and noise. The present study suggests the importance and urgency of accurately encoding the fine-structure cue in cochlear implants. 2005 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

Frequency modulation (FM) detection was investigated in acoustic and electric hearing to characterize cochlear-implant subjects' ability to detect dynamic frequency changes and to assess the relative contributions of temporal and spectral cues to frequency processing. Difference limens were measured for frequency upward sweeps, downward sweeps, and sinusoidal FM as a function of standard frequency and modulation rate. In electric hearing, factors including electrode position and stimulation level were also studied. Electric hearing data showed that the difference limen increased monotonically as a function of standard frequency regardless of the modulation type, the modulation rate, the electrode position, and the stimulation level. In contrast, acoustic hearing data showed that the difference limen was nearly a constant as a function of standard frequency. This difference was interpreted to mean that temporal cues are used only at low standard frequencies and at low modulation rates. At higher standard frequencies and modulation rates, the reliance on the place cue is increased, accounting for the better performance in acoustic hearing than for electric hearing with single-electrode stimulation. The present data suggest a speech processing strategy that encodes slow frequency changes using lower stimulation rates than those typically employed by contemporary cochlear-implant speech processors. [ABSTRACT FROM AUTHOR]

When instructed to speak clearly for people with hearing loss, a talker can effectively enhance the intelligibility of his/her speech by producing "clear" speech. We analyzed global acoustic properties of clear and conversational speech from two talkers and measured their speech intelligibility over a wide range of signal-to-noise ratios in acoustic and electric hearing. Consistent with previous studies, we found that clear speech had a slower overall rate, higher temporal amplitude modulations, and also produced higher intelligibility than conversational speech. To delineate the role of temporal amplitude modulations in clear speech, we extracted the temporal envelope from a number of frequency bands and replaced speech fine-structure with noise fine-structure to simulate cochlear implants. Although both simulated and actual cochlear-implant listeners required higher signal-to-noise ratios to achieve normal performance, a 3-4 dB difference in speech reception threshold was preserved between clear and conversational speech for all experimental conditions. These results suggest that while temporal fine structure is important for speech recognition in noise in general, the temporal envelope carries acoustic cues that contribute to the clear speech intelligibility advantage. [ABSTRACT FROM AUTHOR]

It is important to know what cues the sensory system extracts from natural stimuli and how the brain uses them to form perception. To explore this issue, Smith, Delgutte, and Oxenham [Nature (London) 416, 87-90 (2002)] mixed one sound's temporal envelope with another sound's fine temporal structure to produce auditory chimaeras and found that "the perceptual importance of the envelope increases with the number of frequency bands, while that of the fine structure diminishes." This study addressed two technical issues related to natural cochlear filtering and artificial filter ringing in the chimaerizing algorithm. In addition, this study found that the dichotomy in auditory perception revealed by auditory chimaeras is an epiphenomenon of the classic dichotomy between low- and high-frequency processing. Finally, this study found that the temporal envelope determines sound location as long as the interaural level difference cue is present. The present result reinforces the original hypothesis that the temporal envelope is critical for speech perception whereas temporal fine structure is critical for pitch perception, but does not support the assertion regarding the temporal envelope and fine structure as the acoustic basis for the "what" and "where" mechanisms. [ABSTRACT FROM AUTHOR]

Speech recognition performance was measured in normal-hearing and cochlear-implant listeners with maskers consisting of either steady-state speech-spectrum-shaped noise or a competing sentence. Target sentences from a male talker were presented in the presence of one of three competing talkers (same male, different male, or female! or speech-spectrum-shaped noise generated from this talker at several target-to-masker ratios. For the normal-hearing listeners, target-masker combinations were processed through a noise-excited vocoder designed to simulate a cochlear implant. With unprocessed stimuli, a normal-hearing control group maintained high levels of intelligibility down to target-to-masker ratios as low as 0 dB and showed a release from masking, producing better performance with single-talker maskers than with steady-state noise. In contrast, no masking release was observed in either implant or normal-hearing subjects listening through an implant simulation. The performance of the simulation and implant groups did not improve when the single-talker masker was a different talker compared to the same talker as the target speech, as was found in the normal-hearing control. These results are interpreted as evidence for a significant role of informational masking and modulation interference in cochlear implant speech recognition with fluctuating maskers. This informational masking may originate from increased target-masker similarity when spectral resolution is reduced. [ABSTRACT FROM AUTHOR]

This study examines optimal conversions of speech sounds to audible electric currents in cochlear-implant listeners. The speech dynamic range was measured for 20 consonants and 12 vowels spoken by five female and five male talkers. Even when the maximal root-mean-square (rms) level was normalized for all phoneme tokens, both broadband and narrow-band acoustic analyses showed an approximately 50-dB distribution of speech envelope levels. Phoneme recognition was also obtained in ten CLARION implant users as a function of the input dynamic range from 10 to 80 dB in 10-dB steps. Acoustic amplitudes within a specified input dynamic range were logarithmically mapped into the 10-20-dB range of electric stimulation typically found in cochlear-implant users. Consistent with acoustic data, the perceptual data showed that a 50-60-dB input dynamic range produced optimal speech recognition in these implant users. The present results indicate that speech dynamic range is much greater than the commonly assumed 30-dB range. A new amplitude mapping strategy, based on envelope distribution differences between consonants and vowels, is proposed to optimize acoustic-to-electric mapping of speech sounds. This new strategy will use a logarithmic map for low-frequency channels and a more compressive map for high-frequency channels, and may improve overall speech recognition for cochlear-implant users. © 2002 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

Recognition of consonants, vowels, and sentences was measured in conditions of reduced spectral resolution and distorted spectral distribution of temporal envelope cues. Speech materials were processed through four bandpass filters (analysis bands), half-wave rectified, and low-pass filtered to extract the temporal envelope from each band. The envelope from each speech band modulated a band-limited noise (carrier bands). Analysis and carrier bands were manipulated independently to alter the spectral distribution of envelope cues. Experiment I demonstrated that the location of the cutoff frequencies defining the bands was not a critical parameter for speech recognition, as long as the analysis and carrier bands were matched in frequency extent. Experiment II demonstrated a dramatic decrease in performance when the analysis and carrier bands did not match in frequency extent, which resulted in a warping of the spectral distribution of envelope cues. Experiment III demonstrated a large decrease in performance when the carrier bands were shifted in frequency, mimicking the basal position of electrodes in a cochlear implant. And experiment IV showed a relatively minor effect of the overlap in the noise carrier bands, simulating the overlap in neural populations responding to adjacent electrodes in a cochlear implant. Overall, these results show that, for four bands, the frequency alignment of the analysis bands and carrier bands is critical for good performance, while the exact frequency divisions and overlap in carrier bands are not as critical. © 1998 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

Recent studies have shown that temporal waveform envelope cues can provide significant information for English speech recognition. This study investigated the use of temporal envelope cues in a tonal language: Mandarin Chinese. In this study, the speech was divided into several frequency analysis bands; the amplitude envelope was extracted from each band by half-wave rectification and low-pass filtering and was used to modulate a noise of the same bandwidth as the analysis band. These manipulations preserved temporal and amplitude cues in each frequency band, but removed the spectral detail within each band. Chinese vowels, consonants, tones and sentences were identified by 12 native Chinese-speaking listeners with 1, 2, 3, and 4 noise bands. The results showed that the recognition score of vowels, consonants, and sentences increased monotonically with the number of bands, a pattern similar to that observed in English speech recognition. In contrast, tones were consistently recognized at about 80% correct level, independent of the number of bands. This high level of tone recognition produced a significant difference in the open-set sentence recognition between Chinese (11.0%) and English (2.9%) for the one-band condition where no spectral information was available. The data also revealed that, with primarily temporal cues, the falling-rising tone (tone 3) and the falling tone (tone 4) were more easily recognized than the flat tone (tone 1) and the rising tone (tone 2). This differential pattern in tone recognition resulted in a similar pattern in word recognition: words having either tone 3 or 4 were more likely to be recognized while words having tone 1 and 2 were not. The quantitative role of tones in Chinese speech recognition was further explored using a power-function model and found to play a significant role in relating phoneme recognition to sentence recognition. © 1998 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

Intensity coding mechanisms are explored in a paradigm involving both forward and simultaneous masking. For intensity discrimination of 1000-Hz pure tone in quiet, a near-miss to Weber's law is observed. However, as more stimulus components are added to this relatively simple experiment, interactions among components produce a more complex pattern of results. An intense forward masker, while not causing any threshold shift for the test tone, produces a nonmonotonic intensity discrimination function ['the midlevel hump,' Zeng et al., Hearing Res. 55, 223-230 (1991)]. The midlevel hump can be removed by the presence of additional notched noise [Plack and Viemeister, J. Acoust. Soc. Am. 92, 1902-1910 (1992)] or narrow-band noise whose level is increased along with the test tone's standard level. The same midlevel hump can also be enhanced by a fixed-low-level notched noise or a high-level, high-pass noise which causes minimal masking at the test frequency. Interactions of forward masking and simultaneous masking present a serious problem for a clear interpretation of these results. For example, the notched noise was originally intended to restrict off-frequency listening, but on-frequency masking compromised this original purpose and confounded the interpretation of the notched noise effects. By measuring systematically the growth-of-masking functions, the present study identified various interactions of forward and simultaneous masking and clarified the role of off-frequency listening in forward-masked intensity discrimination. Both peripheral and central mechanisms may have contributed to the occurrence, reduction and enhancement of the midlevel hump under these masking conditions. © 1998 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

Traditional loudness models have been based on the average energy and the critical band analysis of steady-state sounds. However, most environmental sounds, including speech, are dynamic stimuli, in which the average level [e.g., the root-mean-square (rms) level] does not account for the large temporal fluctuations. The question addressed here was whether two stimuli of the same rms level but different peak levels would produce an equal loudness sensation. A modern adaptive procedure was used to replicate two classic experiments demonstrating that the sensation of 'beats' in a two- or three-tone complex resulted in a louder sensation [E. Zwicker and H. Fastl, Psychoacoustics-Facts and Models (Springer-Verlag, Berlin, 1990)]. Two additional experiments were conducted to study exclusively the effects of the temporal envelope on the loudness sensation of dynamic stimuli. Loudness balance was performed by normal-hearing listeners between a white noise and a sinusoidally amplitude-modulated noise in one experiment, and by cochlear implant listeners between two harmonic stimuli of the same magnitude spectra, but different phase spectra, in the other experiment. The results from both experiments showed that, for two stimuli of the same rms level, the stimulus with greater temporal fluctuations sometimes produced a significantly louder sensation, depending on the temporal frequency and overall stimulus level. In normal-hearing listeners, the louder sensation was produced for the amplitude-modulated stimuli with modulation frequencies lower than 400 Hz, and gradually disappeared above 400 Hz, resulting in a low-pass filtering characteristic which bore some similarity to the temporal modulation transfer function. The extent to which loudness was greater was a nonmonotonic function of level in acoustic hearing and a monotonically increasingly function in electric hearing. These results suggest that the loudness sensation of a dynamic stimulus is not limited to a 100-ms temporal integration process, and may be determined jointly by a compression process in the cochlea and an expansion process in the brain. A level-dependent compression scheme that may better restore normal loudness of dynamic stimuli in hearing aids and cochlear implants is proposed. © 1997 Acoustical Society of America. [ABSTRACT FROM AUTHOR]

The growth of loudness of a tone burst following an intense forward masker was measured as a function of the tone level. The level of the forward-masked tone was adjusted to balance the loudness of a standard tone presented without a forward masker, using a 2AFC, double-staircase, tracking procedure. The forward masker was a 90-dB SPL, 100-ms, 1000-Hz pure tone. The standard tone and the masked tone were both 25-ms, 1000-Hz pure tones. The forward masker and the masked tone were always presented in the first interval. With a 100-ms delay between them, there was little or no threshold elevation for the masked tone. However, the masker caused the masked tone to sound louder than it would if it had not been masked, a phenomenon termed ''loudness enhancement'' [Irwin and Zwislocki, Percept. Psychophys. 10, 189-192 (1971); Galambos et al., J. Acoust. Soc. Am. 52, 1127-1130 (1972)]. In addition, the present results show a nonmonotonic enhancement function that the forward masker introduced a 10-16-dB enhancement effect for tones of 40-65 dB SPL and no significant effect for the 30 and 90 dB SPL tones. The loudness variability in forward masking was estimated from the upper and lower sequences tracking the 21% and the 79% louder response levels on the psychometric function, respectively. The variability demonstrated a similar nonmonotonic function. In forward masking loudness grows more steeply at low-medium sensation levels, and merges with normal growth at high levels. The relationship between loudness growth and intensity discrimination is also examined. The present study argues that loudness enhancement does not directly contribute to the midlevel hump on the forward-masked intensity discrimination function, but rather that the increase in loudness variability caused by forward masking is responsible for the midlevel hump. [ABSTRACT FROM AUTHOR]

A nonmonotonic intensity discrimination function was recently reported in which a midlevel hump occurred for 25-ms sinusoidal standards ranging from 20 to 100 dB SPL and presented 100 ms after an intense narrow-band noise forward masker [F.-G. Zeng et al., Hear. Res. 55, 223-230 (1991)]. This paper provides additional data on how the midlevel hump is affected by three factors of forward masking: signal delay, masker level, and frequency. Specifically, just-noticeable differences (jnd's) in intensity were obtained at signal delays of 50, 200, and 400 ms. Results show that at the midlevels the forward-masked intensity jnd's did not recover to their unmasked values, even at the 400-ms signal delay. The longer the signal delay, the smaller this midlevel hump. This slow recovery of the midlevel jnd's is consistent with the finding that low-spontaneous rate (SR) neurons have a slow recovery from forward masking [E. M. Relkin and J. R. Doucet, Hear. Res. 55, 215-222 (1991)]. The large midlevel effect decreased sharply as masker level was reduced from 90 to 60 dB SPL, and disappeared for masker levels less than 40 dB SPL. A frequency selectivity effect for the large midlevel jnd effect was also observed, as maskers with frequency components 2 to 3 oct away from the signal frequency did not affect the jnd's. Overall, the present data are consistent with the hypothesis of Zeng et al. (1991) that low-SR neurons are involved in the midlevel hump of intensity discrimination in forward masking. [ABSTRACT FROM AUTHOR]

Olivocochlear efferents are assumed to be severed during vestibular neurectomy in subjects with severe vertigo. Effects of vestibular neurectomy on pure-tone intensity discrimination and speech perception in noise were examined in six subjects by comparing performance between the surgery ear and the nonsurgery ear, and when available, between the pre- and post-operative conditions. Five of the six subjects had normal or near normal pure-tone average thresholds (<30 dB HL). Broadband white noise was used as in intensity discrimination, and speech spectral-shaped noise was used in speech reception threshold measurements. Both types of noise were presented binaurally at several different levels, whereas the tone or speech was presented monaurally and its level was varied adaptively according to the subject's response. Preliminary results show that (1) loudness dynamic range is not affected by the surgery, (2) intensity and speech perception in noise significantly worsen after the surgery in some subjects but not others. The variability of the results may be due to the fact that not all efferents are severed in every subject during vestibular neurectomy. An objective measure of the efferent nerve survival after vestibular neurectomy in these subjects is needed to further understand perceptual function of the cochlear efferent pathway. [Work supported by NIH-BRSG.] [ABSTRACT FROM AUTHOR]

The growth of loudness as a function of intensity was measured for a brief tone burst following a forward masker. The level of the forward-masked tone was adjusted to balance the loudness of a standard tone presented without a forward masker, using a 2AFC, double-staircase, tracking procedure. The forward masker was a 90 dB SPL, 100-ms, 1000-Hz pure tone. The standard and the balanced stimuli were 25-ms, 1000-Hz tones. The forward masker and the balanced tone were always presented in the first interval and the delay between them was 100 ms. In general, the masker caused the following tone to appear louder than it would if it had not been masked, a phenomenon termed ''loudness enhancement'' [Zwislocki and Sokolich, Percept. Psychophys. 16, 80-90 (1974); Elmasian and Galambos, J. Acoust. Soc. Am. 58, 229-234 (1975)]. In addition, the present results show a nonmonotonic function; a 10- to 20-dB enhancement effect for midlevel tones and no enhancement for low- and high-level tones. The loudness jnd's derived from the upper and lower sequences, also show a similar nonmonotonic function. The present data indicate that loudness grows much steeper following a forward masker and there is a strong relationship between loudness and the jnd's. [Work supported by NIDCD.] [ABSTRACT FROM AUTHOR]

By presenting fricative-vowel tokens across a range of intensities and determining the audibility of various portions of the tokens, the present study investigated the cues for fricative perception. The frication cue was defined as gross spectral-shape differences residing in the steady-state frication noise burst; while the transition cue was defined as dynamic formant changes in the transition portion between the fricative and its adjacent vowel. Both natural and synthetic syllables, with duration and amplitude normalized, were used in closed-set recognition tasks by four normal-hearing subjects. The presentation levels varied from 0%-100% audibility for individual fricatives. Results suggest that the frication burst may be a sufficient cue for the correct perception of fricatives; while the transition cue, although it may not be necessary, plays an important role in helping to identify the place information, particularly at low presentation levels where the frication cue is always not audible. Two other experiments, using truncated stimuli, with frication or transition cues presented independently, further supported our conclusions. [Work supported by Syracuse University Senate Research Grant and Deafness Research Foundation.] [ABSTRACT FROM AUTHOR]

By comparing conditions under which both normal and hearing-impaired subjects were presented with equivalent degrees of audibility for fricative perception, the current study isolated the factors of lack of audibility from that of loss of suprathreshold discriminability of fricative-perception cues. One moderate and two severe cochlear-impaired subjects identified synthetic fricatives, with either full cues (frication and transition) or frication cue alone, presented in closed-set response tasks across a wide range of presentation levels. The results showed that hearing-impaired subjects yielded equivalent recognition performance to the normals when given an equivalent degree of audibility for the frication cue, but they obtained poorer-than-normal performance if only given an equivalent degree of audibility for the transition cue. Considering the 20- to 25-dB intensity difference between the frication and transition cues, the difficulty that hearing-impaired subjects have in perceiving fricatives under normal circumstances may be due to two factors: the lack of audibility in terms of frication cue and the loss of discriminability in terms of transition cue. [Work supported by SU Senate Research Grant and Deafness Research Foundation.] [ABSTRACT FROM AUTHOR]

In cochlear implants loudness is a power function of electrical amplitude at low frequencies (<300 Hz) and an exponential function for higher frequencies [Zeng and Shannon, Science 264, 564-566 (1994)]. In addition, the just-noticeable-difference (jnd) in intensity is inversely proportional to the slope of the loudness function. Implant speech processors generally use low-frequency modulation of a high-frequency carrier. Because the loudness functions are different for the modulator and carrier frequencies, this raises the question: Do the loudness and jnd of a modulated stimulus follow the modulator or the carrier? Intensity discrimination was measured for a 100-Hz sinusoidally modulated 1000-Hz sinusoid or pulse train in implant listeners. The jnd was measured either as an increment in the carrier level for a fixed modulation depth or as an increment in the modulation depth for a fixed carrier level. The results showed that the jnd function of the modulated stimuli is similar to the jnd function of the high-frequency carrier. At high sensation levels, implant listeners can discriminate extremely small changes in modulation depth (1%-2%). This implies that modulated stimuli can produce more jnd steps across the dynamic range than either the modulator or the carrier alone. [ABSTRACT FROM AUTHOR]

Temporal waveform envelope cues provide significant information for English speech recognition, and, when combined with lip reading, could produce near-perfect consonant identification performance [Van Tasell et al., 1152-1161 (1987)]. Tonal patterns are important for Chinese speech recognition and can be effectively conveyed by temporal envelope cues [D. H. Whalen and Y. Xu, Phonetics 49, 25-47 (1992)]. This study investigates whether tones can help Chinese-speaking listeners use envelope cues more effectively than English listeners. The speech envelope was extracted from broad frequency bands and used to modulate a noise of the same bandwidth. Mandarin vowels, consonants, tones, and sentences were identified by ten native Chinese-speaking listeners with 1, 2, 3, and 4 noise bands (or channels). The results showed that recognition of vowels, consonants and sentences increases dramatically with the number of channels, a pattern similar to that observed in English speech recognition. However, tones were consistently recognized at about 80% correct level independent of the number of channels. This high level of tone recognition produced a significant difference in open-set sentence recognition between Chinese (11%) and English (1%, p<0.01) for the one channel condition where no spectral information is available. [ABSTRACT FROM AUTHOR]

A quantitative procedure is derived to evaluate the relative contribution of envelope cues to speech recognition. Recognition data of 16 consonants in the /aCa/ form were collected using signal-correlated noise stimuli in seven normal-hearing listeners. Several distance measures were calculated directly from duration and amplitude of the acoustic envelope. One amplitude distance measure was the Euclidean distance which was computed from the squared difference of the sample-by-sample amplitudes. The second measure was the envelope difference index (EDI) [Fortune et al., Ear Hear. 15, 93-95 (1994)] which was computed from the absolute value of the difference of the sample-by-sample amplitudes. A multidimensional scaling analysis was used to convert the perceptual confusion matrix into a distance matrix and to normalize the different distance measures. Correlation coefficients were computed between the different distance measures and the perceptual data. Preliminary analysis of data from six stop consonants showed that the consonant duration alone is sufficient to explain the perceptual data (r=0.92). Although Euclidean distance conveyed less information (r=0.75) than duration, it was a better measure than the EDI (r=0.31). Evaluation of these measures on the full 16 consonant set will be discussed. [ABSTRACT FROM AUTHOR]

Speech recognition was measured in conditions that systematically reduced the amount of spectral information while preserving temporal envelope information. Speech stimuli were spectrally separated into several frequency bands. The temporal envelope in each band was extracted by half-wave rectification followed by low-pass filtering. Each resulting envelope was then used to modulate a noise band with the same bandwidth and cut-off frequencies as the original analysis band. Identification of consonants (16 consonants in aCa context), vowels (8 vowels in hVd context), and words in simple sentences (CUNY sentences) was measured as a function of the number and frequency distribution of analysis bands, the envelope filter cut-off frequency, and overall spectral shaping. Results as a function of the number of channels show that consonant recognition improves from one to two channels, but less improvement is observed from two to four channels. Vowel recognition improves significantly from one to three channels. Sentence recognition improves with the number of channels, approaching 100% correct with four channels. These results indicate that relatively little spectral detail is sufficient for recognition of speech. Results will be discussed in terms of speech processing strategies for cochlear implants. [Work supported by NIDCD.] [ABSTRACT FROM AUTHOR]

Psychophysical results in gap detection, forward masking, temporal integration, and modulation detection have been obtained in electrically stimulated listeners (both cochlear implant and brain-stem implant) as a function of stimulus intensity. The results indicate that temporal processing is similar in normal-hearing and implant listeners when the intensity scale is normalized. Recent loudness balancing and intensity discrimination experiments have demonstrated that loudness is coded by dual mechanisms depending on stimulus frequency. The loudness of low-frequency (<300 Hz) electrical stimuli is a power-law function of electrical amplitude and the mechanism is inferred to reside in the cochlear nucleus. In contrast, loudness of high-frequency stimuli is an exponential function of electrical amplitude with a more peripheral mechanism. By comparing temporal processing at low and high frequencies the processing order of loudness and timing information in the auditory system can be inferred. [Supported by NIH/NIDCD.] [ABSTRACT FROM AUTHOR]

Plack and Viemeister [J. Acoust. Soc. Am. 92, 1902-1910 (1992)] investigated the role of ''off-frequency'' listening in a nonmonotonic intensity discrimination function under forward masking, in which a midlevel hump was originally reported for 25-ms sinusoidal standards presented 100 ms after an intense masker [Zeng et al., Hear. Res. 55, 223-230 (1991)]. Plack and Viemeister used a notched noise to prevent off-frequency listening and found that the notched noise removed or severely reduced the midlevel hump. A problem with the use of a notched noise in their experiment is that the level of the notched noise increases as a function of the standard level, introducing on-frequency masking in the form of elevated thresholds. They measured intensity discrimination at sensation levels between 10 and 14 dB, independent of the standard level. In the present study, the experiment of Plack and Viemeister was replicated with a narrow-band (800-1200 Hz) noise presented simultaneously with the sinusoidal standards to produce threshold elevations similar to the equivalent notched noise conditions. Presumably, the narrow-band noise does not prevent off-frequency listening, but severely reduces the midlevel hump in a fashion similar to the notched noise. This result suggests that the effects of notched noise on forward-masked intensity discrimination are mainly due to the on-frequency masking. The possible role of excitatory and suppressive masking at the auditory nerve level will be discussed. [Work supported by NIH.] [ABSTRACT FROM AUTHOR]

A two-alternative, two-interval, forced-choice (2AFC) adaptive procedure for subjective judgments is described. This procedure is modified from Jesteadt's widely used procedure [Percept. Psychophys. 28, 85-88 (1980)]. The key modification made in the new procedure is the incorporation of an interchangeable concept that a 250-Hz tone sounds softer than a 1000-Hz tone is equivalent to a 1000-Hz tone sounds louder than a 250-Hz tone. The new procedure is very similar in many respects to a typical adaptive, forced-choice procedure used in measuring the detectability of a signal, and contains the following characteristics: (1) no explicit standard is identified for the subject to match; (2) one simple instruction to the subject and one single type of response from the subject are required (i.e., 'which tone sounds louder?'); (3) a decision rule that yields a direct estimate of the 50% point on a psychometric function is incorporated. The new procedure is validated in the successful measurements of the equal loudness contours and loudness growth functions in normal-hearing and hearing-impaired subjects. [ABSTRACT FROM AUTHOR]

An artificial three-layer neural network is constructed to study possible mechanisms in auditory frequency discrimination. The 110 input units, resembling the peripheral neurons, have two sets of filter shapes: idealized eighth nerve isointensity functions and Patterson-style filters [Moore and Glasberg, Hear. Res. 28, 209-255 (1987)]. Twenty-four intermediate units, similar to critical bands, process information from the input units. The output unit makes a frequency discrimination decision based upon an excitation difference in the intermediate units. Single-band and multiband decision-making rules are tested. When using the idealized eighth nerves as the peripheral filter shape, neither the single-band nor a linear-combination, multiband model satisfactorily predicts experimental data. However, by adjusting the weights from the intermediate units to the decision-making unit by the Delta Rule [Rumelhart and McClelland, Parallel Distributed Processing 1 (MIT, Cambridge, MA, 1988), pp. 444-459], it is found that a weighted-band model can best predict the observed data. The weighted-band model assigns different weights to each band according to its excitation difference: The larger the difference is, the greater the weight. When using Patterson-style filters as the input units, none of the above three models provides a satisfactory prediction. The results suggest that (a) the peripheral filter shape may play a more important role than the central decision rule in frequency discrimination, and (b) the weighted-band model may be a more appropriate central decision rule. [ABSTRACT FROM AUTHOR]

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.

Three experiments were conducted to study relative contributions of speaking rate, temporal envelope, and temporal fine structure to clear speech perception. Experiment I used uniform time scaling to match the speaking rate between clear and conversational speech. Experiment II decreased the speaking rate in conversational speech without processing artifacts by increasing silent gaps between phonetic segments. Experiment III created "auditory chimeras" by mixing the temporal envelope of clear speech with the fine structure of conversational speech, and vice versa. Speech intelligibility in normal-hearing listeners was measured over a wide range of signal-to-noise ratios to derive speech reception thresholds (SRT). The results showed that processing artifacts in uniform time scaling, particularly time compression, reduced speech intelligibility. Inserting gaps in conversational speech improved the SRT by 1.3 dB, but this improvement might be a result of increased short-term signal-to-noise ratios during level normalization. Data from auditory chimeras indicated that the temporal envelope cue contributed more to the clear speech advantage at high signal-to-noise ratios, whereas the temporal fine structure cue contributed more at low signal-to-noise ratios. Taken together, these results suggest that acoustic cues for the clear speech advantage are multiple and distributed.

One reason for the poor pitch performance in current cochlear-implant users may be the highly synchronized neural firing in electric hearing that lacks stochastic properties of neural firing in normal acoustic hearing. This study used three different electric stimulation patterns, jittered, probabilistic, and auditory-model-generated pulses, to mimic some aspects of the normal neural firing pattern in acoustic hearing. Pitch discrimination was measured at standard frequencies of 100, 250, 500, and 1000 Hz on three Nucleus-24 cochlear-implant users. To test the utility of the autocorrelation pitch perception model in electric hearing, one, two, and four electrodes were stimulated independently with the same patterned electric stimulation. Results showed no improvement in performance with any experimental pattern compared to the fixed-rate control. Pitch discrimination was actually worsened with the jittered pattern at low frequencies (125 and 250 Hz) than that of the control, suggesting that externally introduced stochastic properties do not improve pitch perception in electric stimulation. The multiple-electrode stimulation did not improve performance but did not degrade performance either. The present results suggest that both "the right time and the right place" may be needed to restore normal pitch perception in cochlear-implant users.

Speech recognition in noise and music perception is especially challenging for current cochlear implant users. The present study utilizes the residual acoustic hearing in the nonimplanted ear in five cochlear implant users to elucidate the role of temporal fine structure at low frequencies in auditory perception and to test the hypothesis that combined acoustic and electric hearing produces better performance than either mode alone. The first experiment measured speech recognition in the presence of competing noise. It was found that, although the residual low-frequency (<1000 Hz) acoustic hearing produced essentially no recognition for speech recognition in noise, it significantly enhanced performance when combined with the electric hearing. The second experiment measured melody recognition in the same group of subjects and found that, contrary to the speech recognition result, the low-frequency acoustic hearing produced significantly better performance than the electric hearing. It is hypothesized that listeners with combined acoustic and electric hearing might use the correlation between the salient pitch in low-frequency acoustic hearing and the weak pitch in the envelope to enhance segregation between signal and noise. The present study suggests the importance and urgency of accurately encoding the fine-structure cue in cochlear implants.