Effects of Modified Hearing Aid Fittings on Loudness and Tone Quality for Different Acoustic Scenes.

OBJECTIVE: To compare loudness and tone-quality ratings for sounds processed via a simulated five-channel compression hearing aid fitted using NAL-NL2 or using a modification of the fitting designed to be appropriate for the type of listening situation: speech in quiet, speech in noise, music, and noise alone. DESIGN: Ratings of loudness and tone quality were obtained for stimuli presented via a loudspeaker in front of the participant. For normal-hearing participants, levels of 50, 65, and 80 dB SPL were used. For hearing-impaired participants, the stimuli were processed via a simulated hearing aid with five-channel fast-acting compression fitted using NAL-NL2 or using a modified fitting. Input levels to the simulated hearing aid were 50, 65, and 80 dB SPL. All participants listened with one ear plugged. For speech in quiet, the modified fitting was based on the CAM2B method. For speech in noise, the modified fitting used slightly (0 to 2 dB) decreased gains at low frequencies. For music, the modified fitting used increased gains (by 5 to 14 dB) at low frequencies. For noise alone, the modified fitting used decreased gains at all frequencies (by a mean of 1 dB at low frequencies increasing to 8 dB at high frequencies). RESULTS: For speech in quiet, ratings of loudness with the NAL-NL2 fitting were slightly lower than the mean ratings for normal-hearing participants for all levels, while ratings with CAM2B were close to normal for the two lower levels, and slightly greater than normal for the highest level. Ratings of tone quality were close to the optimum value ("just right") for both fittings, except that the CAM2B fitting was rated as very slightly boomy for the 80-dB SPL level. For speech in noise, the ratings of loudness were very close to the normal values and the ratings of tone quality were close to the optimal value for both fittings and for all levels. For music, the ratings of loudness were close to the normal values for NAL-NL2 and slightly above normal for the modified fitting. The tone quality was rated as very slightly tinny for NAL-NL2 and very slightly boomy for the modified fitting. For noise alone, the NAL-NL2 fitting was rated as slightly louder than normal for all levels, while the modified fitting was rated as close to normal. Tone quality was rated as slightly sharper for the NAL-NL2 fitting than for the modified fitting. CONCLUSIONS: Loudness and tone quality can sometimes be made slightly closer to "normal" by modifying gains for different listening situations. The modification for music required to achieve "normal" tone quality appears to be less than used in this study.

Optimal combination of neural temporal envelope and fine structure cues to explain speech identification in background noise.

The dichotomy between acoustic temporal envelope (ENV) and fine structure (TFS) cues has stimulated numerous studies over the past decade to understand the relative role of acoustic ENV and TFS in human speech perception. Such acoustic temporal speech cues produce distinct neural discharge patterns at the level of the auditory nerve, yet little is known about the central neural mechanisms underlying the dichotomy in speech perception between neural ENV and TFS cues. We explored the question of how the peripheral auditory system encodes neural ENV and TFS cues in steady or fluctuating background noise, and how the central auditory system combines these forms of neural information for speech identification. We sought to address this question by (1) measuring sentence identification in background noise for human subjects as a function of the degree of available acoustic TFS information and (2) examining the optimal combination of neural ENV and TFS cues to explain human speech perception performance using computational models of the peripheral auditory system and central neural observers. Speech-identification performance by human subjects decreased as the acoustic TFS information was degraded in the speech signals. The model predictions best matched human performance when a greater emphasis was placed on neural ENV coding rather than neural TFS. However, neural TFS cues were necessary to account for the full effect of background-noise modulations on human speech-identification performance.

Effects of noise reduction on AM perception for hearing-impaired listeners.

Noise reduction (NR) systems are commonplace in modern digital hearing aids. Though not improving speech intelligibility, NR helps the hearing-aid user in terms of lowering noise annoyance, reducing cognitive load and improving ease of listening. Previous psychophysical work has shown that NR does in fact improve the ability of normal-hearing (NH) listeners to discriminate the slow amplitude-modulation (AM) cues representative of those found in speech. The goal of this study was to assess whether this improvement of AM discrimination with NR can also be observed for hearing-impaired (HI) listeners. AM discrimination was measured at two audio frequencies of 500 Hz and 2 kHz in a background noise with a signal-to-noise ratio of 12 dB. Discrimination was measured for ten HI and ten NH listeners with and without NR processing. The HI listeners had a moderate sensorineural hearing loss of about 50 dB HL at 2 kHz and normal hearing (≤ 20 dB HL) at 500 Hz. The results showed that most of the HI listeners tended to benefit from NR at 500 Hz but not at 2 kHz. However, statistical analyses showed that HI listeners did not benefit significantly from NR at any frequency region. In comparison, the NH listeners showed a significant benefit from NR at both frequencies. For each condition, the fidelity of AM transmission was quantified by a computational model of early auditory processing. The parameters of the model were adjusted separately for the two groups (NH and HI) of listeners. The AM discrimination performance of the HI group (with and without NR) was best captured by a model simulating the loss of the fast-acting amplitude compression applied by the normal cochlea. This suggests that the lack of benefit from NR for HI listeners results from loudness recruitment.

Modelling the distortion produced by cochlear compression.

Lyon (J Acoust Soc Am 130:3893-3904, 2011) has described how a cascade of simple asymmetric resonators (CAR) can be used to simulate the filtering of the basilar membrane and how the gain of the resonators can be manipulated by a feedback network to simulate the fast-acting compression (FAC) characteristic of cochlear processing. When the compression is applied to complex tones, each pair of primary components produces both quadratic and cubic distortion tones (DTs), and the cascade architecture of the CAR-FAC system propagates them down to their appropriate place along the basilar membrane, where they combine additively with each other and any primary components at that frequency. This suggests that CAR-FAC systems might be used to study the role of compressive distortion in the perception of complex sounds and that behavioural measurements of cochlear distortion data might be useful when tuning the parameters of CAR-FAC systems.

Effects of noise reduction on AM and FM perception.

The goal of noise reduction (NR) algorithms in digital hearing aid devices is to reduce background noise whilst preserving as much of the original signal as possible. These algorithms may increase the signal-to-noise ratio (SNR) in an ideal case, but they generally fail to improve speech intelligibility. However, due to the complex nature of speech, it is difficult to disentangle the numerous low- and high-level effects of NR that may underlie the lack of speech perception benefits. The goal of this study was to better understand why NR algorithms do not improve speech intelligibility by investigating the effects of NR on the ability to discriminate two basic acoustic features, namely amplitude modulation (AM) and frequency modulation (FM) cues, known to be crucial for speech identification in quiet and in noise. Here, discrimination of complex, non-linguistic AM and FM patterns was measured for normal hearing listeners using a same/different task. The stimuli were generated by modulating 1-kHz pure tones by either a two-component AM or FM modulator with patterns changed by manipulating component phases. Modulation rates were centered on 3 Hz. Discrimination of AM and FM patterns was measured in quiet and in the presence of a white noise that had been passed through a gammatone filter centered on 1 kHz. The noise was presented at SNRs ranging from -6 to +12 dB. Stimuli were left as such or processed via an NR algorithm based on the spectral subtraction method. NR was found to yield small but systematic improvements in discrimination for the AM conditions at favorable SNRs but had little effect, if any, on FM discrimination. A computational model of early auditory processing was developed to quantify the fidelity of AM and FM transmission. The model captured the improvement in discrimination performance for AM stimuli at high SNRs with NR. However, the model also predicted a relatively small detrimental effect of NR for FM stimuli in contrast with the average psychophysical data. Overall, these results suggest that the lack of benefits of NR on speech intelligibility is partly caused by the limited effect of NR on the transmission of narrowband speech modulation cues.

Age-related difference in melodic pitch perception is probably mediated by temporal processing: empirical and computational evidence.

OBJECTIVE: This study was designed to examine whether age-related differences in melodic pitch perception may be mediated by temporal processing. Temporal models of pitch suggest that performance will decline as the lowest component of a complex tone increases in frequency, regardless of age. In addition, if there are age-related deficits in temporal processing in older adults, this group may have reduced performance relative to younger adults even in the most favorable conditions. DESIGN: Six younger adults and 10 older adults with clinically normal audiograms up to 8 kHz were tested in a melodic pitch perception task. In each trial, two consecutive four-note melodies were presented to the listener. Melodies were identical with the exception of one note in the second melody that was shifted in pitch. The listener was required to identify which note was shifted. All notes consisted of eight successive harmonic components, with the average lowest component manipulated to be the 4th, 8th, or 12th component of the harmonic series, with lower components being absent. RESULTS: Age-related differences in melodic pitch perception were only apparent when stimulus parameters favored temporal processing of pitch. Furthermore, modeling a loss of periodicity coding yielded an outcome consistent with the observed behavioral results. Although younger adults generally outperformed older adults, about one-quarter of the older adults performed at levels that were equivalent to those of younger adults. The only follow-up tests that were able to differentiate these exceptional older adults were tests that would be sensitive to temporal processing: fundamental frequency difference limens and 500 Hz pure-tone difference limens. In contrast, otoacoustic emissions and high-frequency pure-tone thresholds, which are more commonly associated with spectral processing deficits, were not able to differentiate older exceptional adults from older typical adults. CONCLUSION: Age-related declines in temporal processing contribute to deficits in melodic pitch perception. However, some exceptional older adults with normal audiograms preserve excellent temporal processing and continue to perform at levels that are typical of younger adults.

Location and acoustic scale cues in concurrent speech recognition.

Location and acoustic scale cues have both been shown to have an effect on the recognition of speech in multi-speaker environments. This study examines the interaction of these variables. Subjects were presented with concurrent triplets of syllables from a target voice and a distracting voice, and asked to recognize a specific target syllable. The task was made more or less difficult by changing (a) the location of the distracting speaker, (b) the scale difference between the two speakers, and/or (c) the relative level of the two speakers. Scale differences were produced by changing the vocal tract length and glottal pulse rate during syllable synthesis: 32 acoustic scale differences were used. Location cues were produced by convolving head-related transfer functions with the stimulus. The angle between the target speaker and the distracter was 0 degrees, 4 degrees, 8 degrees, 16 degrees, or 32 degrees on the 0 degrees horizontal plane. The relative level of the target to the distracter was 0 or -6 dB. The results show that location and scale difference interact, and the interaction is greatest when one of these cues is small. Increasing either the acoustic scale or the angle between target and distracter speakers quickly elevates performance to ceiling levels.

Pitch strength decreases as F0 and harmonic resolution increase in complex tones composed exclusively of high harmonics.

A melodic pitch experiment was performed to demonstrate the importance of time-interval resolution for pitch strength. The experiments show that notes with a low fundamental (75 Hz) and relatively few resolved harmonics support better performance than comparable notes with a higher fundamental (300 Hz) and more resolved harmonics. Two four note melodies were presented to listeners and one note in the second melody was changed by one or two semitones. Listeners were required to identify the note that changed. There were three orthogonal stimulus dimensions: F0 (75 and 300 Hz); lowest frequency component (3, 7, 11, or 15); and number of harmonics (4 or 8). Performance decreased as the frequency of the lowest component increased for both F0's, but performance was better for the lower F0. The spectral and temporal information in the stimuli were compared using a time-domain model of auditory perception. It is argued that the distribution of time intervals in the auditory nerve can explain the decrease in performance as F0, and spectral resolution increase. Excitation patterns based on the same time-interval information do not contain sufficient resolution to explain listener's performance on the melody task.

Neural representation of auditory size in the human voice and in sounds from other resonant sources.

The size of a resonant source can be estimated by the acoustic-scale information in the sound [1-3]. Previous studies revealed that posterior superior temporal gyrus (STG) responds to acoustic scale in human speech when it is controlled for spectral-envelope change (unpublished data). Here we investigate whether the STG activity is specific to the processing of acoustic scale in human voice or whether it reflects a generic mechanism for the analysis of acoustic scale in resonant sources. In two functional magnetic resonance imaging (fMRI) experiments, we measured brain activity in response to changes in acoustic scale in different categories of resonant sound (human voice, animal call, and musical instrument). We show that STG is activated bilaterally for spectral-envelope changes in general; it responds to changes in category as well as acoustic scale. Activity in left posterior STG is specific to acoustic scale in human voices and not responsive to acoustic scale in other resonant sources. In contrast, the anterior temporal lobe and intraparietal sulcus are activated by changes in acoustic scale across categories. The results imply that the human voice requires special processing of acoustic scale, whereas the anterior temporal lobe and intraparietal sulcus process auditory size information independent of source category.

Discrimination of speaker size from syllable phrases.

The length of the vocal tract is correlated with speaker size and, so, speech sounds have information about the size of the speaker in a form that is interpretable by the listener. A wide range of different vocal tract lengths exist in the population and humans are able to distinguish speaker size from the speech. Smith et al. [J. Acoust. Soc. Am. 117, 305-318 (2005)] presented vowel sounds to listeners and showed that the ability to discriminate speaker size extends beyond the normal range of speaker sizes which suggests that information about the size and shape of the vocal tract is segregated automatically at an early stage in the processing. This paper reports an extension of the size discrimination research using a much larger set of speech sounds, namely, 180 consonant-vowel and vowel-consonant syllables. Despite the pronounced increase in stimulus variability, there was actually an improvement in discrimination performance over that supported by vowel sounds alone. Performance with vowel-consonant syllables was slightly better than with consonant-vowel syllables. These results support the hypothesis that information about the length of the vocal tract is segregated at an early stage in auditory processing.