Adaptive bandwidth measurements of importance functions for speech intelligibility prediction

The Articulation Index (AI) and Speech Intelligibility Index (SII) predict intelligibility scores from measurements of speech and hearing parameters. One component in the prediction is the "importance function," a weighting function that characterizes contributions of particular spectral regions of speech to speech intelligibility. Previous work with SII predictions for hearing-impaired subjects suggests that prediction accuracy might improve if importance functions for individual subjects were available. Unfortunately, previous importance function measurements have required extensive intelligibility testing with groups of subjects, using speech processed by various fixed-bandwidth low-pass and high-pass filters. A more efficient approach appropriate to individual subjects is desired. The purpose of this study was to evaluate the feasibility of measuring importance functions for individual subjects with adaptive-bandwidth filters. In two experiments, ten subjects with normal-hearing listened to vowel-consonant-vowel (VCV) nonsense words processed by low-pass and high-pass filters whose bandwidths were varied adaptively to produce specified performance levels in accordance with the transformed up-down rules of Levitt [(1971). J. Acoust. Soc. Am. 49, 467-477]. Local linear psychometric functions were fit to resulting data and used to generate an importance function for VCV words. Results indicate that the adaptive method is reliable and efficient, and produces importance function data consistent with that of the corresponding AI/SII importance function.     

Algorithms for separating the speech of interfering talkers: evaluations with voiced sentences, and normal-hearing and hearing-impaired listeners

Two signal-processing algorithms, derived from those described by Stubbs and Summerfield [R.J. Stubbs and Q. Summerfield, J. Acoust. Soc. Am. 84, 1236-1249 (1988)], were used to separate the voiced speech of two talkers speaking simultaneously, at similar intensities, in a single channel. Both algorithms use fundamental frequency (FO) as the basis for segregation. One attenuates the interfering voice by filtering the cepstrum of the signal. The other is a hybrid algorithm that combines cepstral filtering with the technique of harmonic selection [T.W. Parsons, J. Acoust. Soc. Am. 60, 911-918 (1976)]. The algorithms were evaluated and compared in perceptual experiments involving listeners with normal hearing and listeners with cochlear hearing impairments. In experiment 1 the processing was used to separate voiced sentences spoken on a monotone. Both algorithms gave significant increases in intelligibility to both groups of listeners. The improvements were equivalent to an increase of 3-4 dB in the effective signal-to-noise ratio (SNR). In experiment 2 the processing was used to separate voiced sentences spoken with time-varying intonation. For normal-hearing listeners, cepstral filtering gave a significant increase in intelligibility, while the hybrid algorithm gave an increase that was on the margins of significance (p = 0.06). The improvements were equivalent to an increase of 2-3 dB in the effective SNR. For impaired listeners, no intelligibility improvements were demonstrated with intoned sentences. The decrease in performance for intoned material is attributed to limitations of the algorithms when FO is nonstationary.     

Application of an extended equalization-cancellation model to speech intelligibility with spatially distributed maskers

An extended version of the equalization-cancellation (EC) model of binaural processing is described and applied to speech intelligibility tasks in the presence of multiple maskers. The model incorporates time-varying jitters, both in time and amplitude, and implements the equalization and cancellation operations in each frequency band independently. The model is consistent with the original EC model in predicting tone-detection performance for a large set of configurations. When the model is applied to speech, the speech intelligibility index is used to predict speech intelligibility performance in a variety of conditions. Specific conditions addressed include different types of maskers, different numbers of maskers, and different spatial locations of maskers. Model predictions are compared with empirical measurements reported by Hawley et al. [J. Acoust. Soc. Am. 115, 833-843 (2004)] and by Marrone et al. [J. Acoust. Soc. Am. 124, 1146-1158 (2008)]. The model succeeds in predicting speech intelligibility performance when maskers are speech-shaped noise or broadband-modulated speech-shaped noise but fails when the maskers are speech or reversed speech.     

Characterizing auditory processing and perception in individual listeners with sensorineural hearing loss

This study considered consequences of sensorineural hearing loss in ten listeners. The characterization of individual hearing loss was based on psychoacoustic data addressing audiometric pure-tone sensitivity, cochlear compression, frequency selectivity, temporal resolution, and intensity discrimination. In the experiments it was found that listeners with comparable audiograms can show very different results in the supra-threshold measures. In an attempt to account for the observed individual data, a model of auditory signal processing and perception [Jepsen et al., J. Acoust. Soc. Am. 124, 422-438 (2008)] was used as a framework. The parameters of the cochlear processing stage of the model were adjusted to account for behaviorally estimated individual basilar-membrane input-output functions and the audiogram, from which the amounts of inner hair-cell and outer hair-cell losses were estimated as a function of frequency. All other model parameters were left unchanged. The predictions showed a reasonably good agreement with the measured individual data in the frequency selectivity and forward masking conditions while the variation of intensity discrimination thresholds across listeners was underestimated by the model. The model and the associated parameters for individual hearing-impaired listeners might be useful for investigating effects of individual hearing impairment in more complex conditions, such as speech intelligibility in noise.     

Development of the Cantonese speech intelligibility index

A Speech Intelligibility Index (SII) for the sentences in the Cantonese version of the Hearing In Noise Test (CHINT) was derived using conventional procedures described previously in studies such as Studebaker and Sherbecoe [J. Speech Hear. Res. 34, 427-438 (1991)]. Two studies were conducted to determine the signal-to-noise ratios and high- and low-pass filtering conditions that should be used and to measure speech intelligibility in these conditions. Normal hearing subjects listened to the sentences presented in speech-spectrum shaped noise. Compared to other English speech assessment materials such as the English Hearing In Noise Test [Nilsson et al., J. Acoust. Soc. Am. 95, 1085-1099 (1994)], the frequency importance function of the CHINT suggests that low-frequency information is more important for Cantonese speech understanding. The difference in ,frequency importance weight in Chinese, compared to English, was attributed to the redundancy of test material, tonal nature of the Cantonese language, or a combination of these factors.     

Acoustic properties of naturally produced clear speech at normal speaking rates

Sentences spoken "clearly" are significantly more intelligible than those spoken "conversationally" for hearing-impaired listeners in a variety of backgrounds [Picheny et al., J. Speech Hear. Res. 28, 96-103 (1985); Uchanski et al., ibid. 39, 494-509 (1996); Payton et al., J. Acoust. Soc. Am. 95, 1581-1592 (1994)]. While producing clear speech, however, talkers often reduce their speaking rate significantly [Picheny et al., J. Speech Hear. Res. 29, 434-446 (1986); Uchanski et al., ibid. 39, 494-509 (1996)]. Yet speaking slowly is not solely responsible for the intelligibility benefit of clear speech (over conversational speech), since a recent study [Krause and Braida, J. Acoust. Soc. Am. 112, 2165-2172 (2002)] showed that talkers can produce clear speech at normal rates with training. This finding suggests that clear speech has inherent acoustic properties, independent of rate, that contribute to improved intelligibility. Identifying these acoustic properties could lead to improved signal processing schemes for hearing aids. To gain insight into these acoustical properties, conversational and clear speech produced at normal speaking rates were analyzed at three levels of detail (global, phonological, and phonetic). Although results suggest that talkers may have employed different strategies to achieve clear speech at normal rates, two global-level properties were identified that appear likely to be linked to the improvements in intelligibility provided by clear/normal speech: increased energy in the 1000-3000-Hz range of long-term spectra and increased modulation depth of low frequency modulations of the intensity envelope. Other phonological and phonetic differences associated with clear/normal speech include changes in (1) frequency of stop burst releases, (2) VOT of word-initial voiceless stop consonants, and (3) short-term vowel spectra.     

Short-term adaptation to accented English by younger and older adults

This study examined the effects of age and hearing loss on short-term adaptation to accented speech. Data from younger and older listeners in a prior investigation [Gordon-Salant et al. (2010). J. Acoust. Soc. Am. 128, 444-455] were re-analyzed to examine changes in recognition over four administrations of equivalent lists of English stimuli recorded by native speakers of Spanish and English. Results showed improvement in recognition scores over four list administrations for the accented stimuli but not for the native English stimuli. Group effects emerged but were not involved in any interactions, suggesting that short-term adaptation to accented speech is preserved with aging and with hearing loss.     

Use of intonation contours for speech recognition in noise by cochlear implant recipients

The corruption of intonation contours has detrimental effects on sentence-based speech recognition in normal-hearing listeners Binns and Culling [(2007). J. Acoust. Soc. Am. 122, 1765-1776]. This paper examines whether this finding also applies to cochlear implant (CI) recipients. The subjects' F0-discrimination and speech perception in the presence of noise were measured, using sentences with regular and inverted F0-contours. The results revealed that speech recognition for regular contours was significantly better than for inverted contours. This difference was related to the subjects' F0-discrimination providing further evidence that the perception of intonation patterns is important for the CI-mediated speech recognition in noise.     

Prediction of the influence of reverberation on binaural speech intelligibility in noise and in quiet

Reverberation usually degrades speech intelligibility for spatially separated speech and noise sources since spatial unmasking is reduced and late reflections decrease the fidelity of the received speech signal. The latter effect could not satisfactorily be predicted by a recently presented binaural speech intelligibility model [Beutelmann et al. (2010). J. Acoust. Soc. Am. 127, 2479-2497]. This study therefore evaluated three extensions of the model to improve its predictions: (1) an extension of the speech intelligibility index based on modulation transfer functions, (2) a correction factor based on the room acoustical quantity "definition," and (3) a separation of the speech signal into useful and detrimental parts. The predictions were compared to results of two experiments in which speech reception thresholds were measured in a reverberant room in quiet and in the presence of a noise source for listeners with normal hearing. All extensions yielded better predictions than the original model when the influence of reverberation was strong, while predictions were similar for conditions with less reverberation. Although model (3) differed substantially in the assumed interaction of binaural processing and early reflections, its predictions were very similar to model (2) that achieved the best fit to the data.     

Effects of noise and distortion on speech quality judgments in normal-hearing and hearing-impaired listeners

Noise and distortion reduce speech intelligibility and quality in audio devices such as hearing aids. This study investigates the perception and prediction of sound quality by both normal-hearing and hearing-impaired subjects for conditions of noise and distortion related to those found in hearing aids. Stimuli were sentences subjected to three kinds of distortion (additive noise, peak clipping, and center clipping), with eight levels of degradation for each distortion type. The subjects performed paired comparisons for all possible pairs of 24 conditions. A one-dimensional coherence-based metric was used to analyze the quality judgments. This metric was an extension of a speech intelligibility metric presented in Kates and Arehart (2005) [J. Acoust. Soc. Am. 117, 2224-2237] and is based on dividing the speech signal into three amplitude regions, computing the coherence for each region, and then combining the three coherence values across frequency in a calculation based on the speech intelligibility index. The one-dimensional metric accurately predicted the quality judgments of normal-hearing listeners and listeners with mild-to-moderate hearing loss, although some systematic errors were present. A multidimensional analysis indicates that several dimensions are needed to describe the factors used by subjects to judge the effects of the three distortion types.     

Similarity in loudness and distortion product otoacoustic emission input/output functions: implications for an objective hearing aid adjustment

The aim of the present study was to compare distortion product otoacoustic emissions (DPOAEs) to loudness with regard to the potentiality of DPOAEs to determine characteristic quantities of the cochlear-impaired ear and to derive objective hearing aid parameters. Recently, Neely et al. [J. Acoust. Soc. Am. 114, 1499-1507 (2003)] compared DPOAE input/output functions to the Fletcher and Munson [J. Acoust. Soc. Am. 5, 82-108 (1933)] loudness function finding a close resemblance in the slope characteristics of both measures. The present study extended their work by performing both loudness and DPOAE measurements in the same subject sample, and by developing a method for the estimation of gain needed to compensate for loss of cochlear sensitivity and compression. DPOAEs and loudness exhibited similar behavior when plotted on a logarithmic scale and slope increased with increasing hearing loss, confirming the findings of Neely et al. To compensate for undesired nonpathological impacts on the magnitude of DPOAE level, normalization of DPOAE data was implemented. A close resemblance between gain functions based on loudness and normalized DPOAE data was achieved. These findings suggest that DPOAEs are able to quantify the loss of cochlear sensitivity and compression and thus might provide parameters for a noncooperative hearing aid adjustment.     

The effects of hearing loss on the contribution of high- and low-frequency speech information to speech understanding. II. Sloping hearing loss

The speech understanding of persons with sloping high-frequency (HF) hearing impairment (HI) was compared to normal hearing (NH) controls and previous research on persons with "flat" losses [Hornsby and Ricketts (2003). J. Acoust. Soc. Am. 113, 1706-1717] to examine how hearing loss configuration affects the contribution of speech information in various frequency regions. Speech understanding was assessed at multiple low- and high-pass filter cutoff frequencies. Crossover frequencies, defined as the cutoff frequencies at which low- and high-pass filtering yielded equivalent performance, were significantly lower for the sloping HI, compared to NH, group suggesting that HF HI limits the utility of HF speech information. Speech intelligibility index calculations suggest this limited utility was not due simply to reduced audibility but also to the negative effects of high presentation levels and a poorer-than-normal use of speech information in the frequency region with the greatest hearing loss (the HF regions). This deficit was comparable, however, to that seen in low-frequency regions of persons with similar HF thresholds and "flat" hearing losses suggesting that sensorineural HI results in a "uniform," rather than frequency-specific, deficit in speech understanding, at least for persons with HF thresholds up to 60-80 dB HL.     

Effects of signal-to-noise ratio, signal periodicity, and degree of hearing impairment on the performance of voice-separation algorithms

Procedures for enhancing the intelligibility of a target talker in the presence of a co-channel competing talker were evaluated in tests involving (i) continuously voiced sentences spoken on a monotone, (ii) continuously voiced sentences with time-varying intonation, and (iii) noncontinuously voiced sentences produced with natural intonation. The procedures were based on the methods of harmonic selection and cepstral filtering [R.J. Stubbs and Q. Summerfield, J. Acoust. Soc. Am. 87, 359-372 (1990)]. Target and competing voices were combined at signal-to-noise ratios (SNRs) between -10 dB and +10 dB. Subjects were a group with normal hearing and a heterogeneous group with mild-moderate cochlear hearing impairments. Processing enhanced the target voice over a range of SNRs for each type of sentence and for most listeners. Enhancement was greatest at negative SNRs. Among the impaired listeners, benefit was generally greater for those with milder losses. These results consolidate and extend previous demonstrations that voice-separation algorithms that exploit the harmonic structure of the voiced portions of speech can enhance intelligibility. However, practical application of such algorithms depends on a solution to the problem of tracking the fundamental-frequency contour of one voice in the presence of a competing voice.     

Effects of perilymph viscosity on low-frequency intracochlear pressures and the cochlear input impedance of the cat

Cochlear model calculations are shown to be in reasonable agreement with recent low-frequency measurements of intracochlear pressures and the cochlear input impedance of the cat [V. Nedzelnitsky, J. Acoust. Soc. Am. 68, 1676-1689 (1980); T. J. Lynch, III, V. Nedzelnitsky, and W. T. Peake, J. Acoust. Soc. Am. 72, 108-130 (1982)]. Included in the cochlear model are perilymph viscosity, the measured variation of the area of the scala vestibuli with distance from the stapes [P. Dallos, J. Acoust. Soc. Am. 48, 489-499 (1970)], and finite impedance of the round window membrane. The WKB approximation and its extension to the low-frequency region is used in order to exhibit explicitly the dependence of the model results on the cochlear parameters.     

Stop-consonant recognition for normal-hearing listeners and listeners with high-frequency hearing loss. II: Articulation index predictions

Articulation index (AI) theory was used to evaluate stop-consonant recognition of normal-hearing listeners and listeners with high-frequency hearing loss. From results reported in a companion article [Dubno et al., J. Acoust. Soc. Am. 85, 347-354 (1989)], a transfer function relating the AI to stop-consonant recognition was established, and a frequency importance function was determined for the nine stop-consonant-vowel syllables used as test stimuli. The calculations included the rms and peak levels of the speech that had been measured in 1/3 octave bands; the internal noise was estimated from the thresholds for each subject. The AI model was then used to predict performance for the hearing-impaired listeners. A majority of the AI predictions for the hearing-impaired subjects fell within +/- 2 standard deviations of the normal-hearing listeners' results. However, as observed in previous data, the AI tended to overestimate performance of the hearing-impaired listeners. The accuracy of the predictions decreased with the magnitude of high-frequency hearing loss. Thus, with the exception of performance for listeners with severe high-frequency hearing loss, the results suggest that poorer speech recognition among hearing-impaired listeners results from reduced audibility within critical spectral regions of the speech stimuli.     

Perceptual contributions to monosyllabic word intelligibility: segmental, lexical, and noise replacement factors

This study investigated the relative contributions of consonants and vowels to the perceptual intelligibility of monosyllabic consonant-vowel-consonant (CVC) words. A noise replacement paradigm presented CVCs with only consonants or only vowels preserved. Results demonstrated no difference between overall word accuracy in these conditions; however, different error patterns were observed. A significant effect of lexical difficulty was demonstrated for both types of replacement, whereas the noise level used during replacement did not influence results. The contribution of consonant and vowel transitional information present at the consonant-vowel boundary was also explored. The proportion of speech presented, regardless of the segmental condition, overwhelmingly predicted performance. Comparisons were made with previous segment replacement results using sentences [Fogerty, and Kewley-Port (2009). J. Acoust. Soc. Am. 126, 847-857]. Results demonstrated that consonants contribute to intelligibility equally in both isolated CVC words and sentences. However, vowel contributions were mediated by context, with greater contributions to intelligibility in sentence contexts. Therefore, it appears that vowels in sentences carry unique speech cues that greatly facilitate intelligibility which are not informative and/or present during isolated word contexts. Consonants appear to provide speech cues that are equally available and informative during sentence and isolated word presentations.     

Frequency importance functions for a feature recognition test material

The relative importance of different parts of the auditory spectrum to recognition of the Diagnostic Rhyme Test (DRT) and its six speech feature subtests was determined. Three normal hearing subjects were tested twice in each of 70 experimental conditions. The analytical procedures of French and Steinberg [J. Acoust. Soc. Am. 19, 90-119 (1947)] were applied to the data to derive frequency importance functions for each of the DRT subtests and the test as a whole over the frequency range 178-8912 Hz. For the DRT as a whole, the low frequencies were found to be more important than is the case for nonsense syllables. Importance functions for the feature subtests also differed from those for nonsense syllables and from each other as well. These results suggest that test materials loaded with different proportions of particular phonemes have different frequency importance functions. Comparison of the results with those from other studies suggests that importance functions depend to a degree on the available response options as well.     

An articulation index based procedure for predicting the speech recognition performance of hearing-impaired individuals

An articulation index calculation procedure developed for use with individual normal-hearing listeners [C. Pavlovic and G. Studebaker, J. Acoust. Soc. Am. 75, 1606-1612 (1984)] was modified to account for the deterioration in suprathreshold speech processing produced by sensorineural hearing impairment. Data from four normal-hearing and four hearing-impaired subjects were used to relate the loss in hearing sensitivity to the deterioration in speech processing in quiet and in noise. The new procedure only requires hearing threshold measurements and consists of the following two modifications of the original AI procedure of Pavlovic and Studebaker (1984): The speech and noise spectrum densities are integrated over bandwidths which are, when expressed in decibels, larger than the critical bandwidths by 10% of the hearing loss. This is in contrast to the unmodified procedure where integration is performed over critical bandwidths. The contribution of each frequency to the AI is the product of its contribution in the unmodified AI procedure and a "speech desensitization factor." The desensitization factor is specified as a function of the hearing loss. The predictive accuracies of both the unmodified and the modified calculation procedures were assessed by comparing the expected and observed speech recognition scores of four hearing-impaired subjects under various conditions of speech filtering and noise masking. The modified procedure appears accurate for general applications. In contrast, the unmodified procedure appears accurate only for applications where results obtained under various conditions on a single listener are compared to each other.     

Improving syllable identification by a preprocessing method reducing overlap-masking in reverberant environments

Overlap-masking degrades speech intelligibility in reverberation [R. H. Bolt and A. D. MacDonald, J. Acoust. Soc. Am. 21(6), 577-580 (1949)]. To reduce the effect of this degradation, steady-state suppression has been proposed as a preprocessing technique [Arai et al., Proc. Autumn Meet. Acoust. Soc. Jpn., 2001; Acoust. Sci. Tech. 23(8), 229-232 (2002)]. This technique automatically suppresses steady-state portions of speech that have more energy but are less crucial for speech perception. The present paper explores the effect of steady-state suppression on syllable identification preceded by /a/ under various reverberant conditions. In each of two perception experiments, stimuli were presented to 22 subjects with normal hearing. The stimuli consisted of mono-syllables in a carrier phrase with and without steady-state suppression and were presented under different reverberant conditions using artificial impulse responses. The results indicate that steady-state suppression statistically improves consonant identification for reverberation times of 0.7 to 1.2 s. Analysis of confusion matrices shows that identification of voiced consonants, stop and nasal consonants, and bilabial, alveolar, and velar consonants were especially improved by steady-state suppression. The steady-state suppression is demonstrated to be an effective preprocessing method for improving syllable identification by reducing the effect of overlap-masking under specific reverberant conditions.     

Illusory spectrotemporal ripples created with binaurally correlated noise

Binaural disparities are the primary acoustic cues employed in sound localization tasks. However, the degree of binaural correlation in a sound serves as a complementary cue for detecting competing sound sources [J. F. Culling, H. S. Colburn, and M. Spurchise, "Interaural correlation sensitivity," J. Acoust. Soc. Am. 110(2), 1020-1029 (2001) and L. R. Bernstein and C. Trahiotis, "On the use of the normalized correlation as an index of interaural envelope correlation," J. Acoust. Soc. Am. 100, 1754-1763 (1996)]. Here a random chord stereogram (RCS) sound is developed that produces a salient pop-out illusion of a slowly varying ripple sound [T. Chi et al., "Spectro-temporal modulation transfer functions and speech intelligibility," J. Acoust. Soc. Am. 106(5), 2719-2732 (1999)], even though the left and right ear sounds alone consist of noise-like random modulations. The quality and resolution of this percept is systematically controlled by adjusting the spectrotemporal correlation pattern between the left and right sounds. The prominence and limited time-frequency resolution for resolving the RCS suggests that envelope correlations are a dominant binaural cue for grouping acoustic objects.