Effect of noise and occupancy on optimal reverberation times for speech intelligibility in classrooms

The question of what is the optimal reverberation time for speech intelligibility in an occupied classroom has been studied recently in two different ways, with contradictory results. Experiments have been performed under various conditions of speech-signal to background-noise level difference and reverberation time, finding an optimal reverberation time of zero. Theoretical predictions of appropriate speech-intelligibility metrics, based on diffuse-field theory, found nonzero optimal reverberation times. These two contradictory results are explained by the different ways in which the two methods account for background noise, both of which are unrealistic. To obtain more realistic and accurate predictions, noise sources inside the classroom are considered. A more realistic treatment of noise is incorporated into diffuse-field theory by considering both speech and noise sources and the effects of reverberation on their steady-state levels. The model shows that the optimal reverberation time is zero when the speech source is closer to the listener than the noise source, and nonzero when the noise source is closer than the speech source. Diffuse-field theory is used to determine optimal reverberation times in unoccupied classrooms given optimal values for the occupied classroom. Resulting times can be as high as several seconds in large classrooms; in some cases, optimal values are unachievable, because the occupants contribute too much absorption.     

Relations among temporal acuity, hearing loss, and the perception of speech distorted by noise and reverberation

Eight normal listeners and eight listeners with sensorineural hearing losses were compared on a gap-detection task and on a speech perception task. The minimum detectable gap (71% correct) was determined as a function of noise level, and a time constant was computed from these data for each listener. The time constants of the hearing-impaired listeners were significantly longer than those of the normal listeners. The speech consisted of sentences that were mixed with two levels of noise and subjected to two kinds of reverberation (real or simulated). The speech thresholds (minimum signal-to-noise ratio for 50% correct) were significantly higher for the hearing-impaired listeners than for the normal listeners for both kinds of reverberation. The longer reverberation times produced significantly higher thresholds than the shorter times. The time constant was significantly correlated with all the speech threshold measures (r = -0.58 to -0.74) and a measure of hearing threshold loss also correlated significantly with all the speech thresholds (r = 0.53 to 0.95). A principal components analysis yielded two factors that accounted for the intercorrelations. The factor loadings for the time constant were similar to those on the speech thresholds for real reverberation and the loadings for hearing loss were similar to those of the thresholds for simulated reverberation.     

Integration efficiency for speech perception within and across sensory modalities by normal-hearing and hearing-impaired individuals

In face-to-face speech communication, the listener extracts and integrates information from the acoustic and optic speech signals. Integration occurs within the auditory modality (i.e., across the acoustic frequency spectrum) and across sensory modalities (i.e., across the acoustic and optic signals). The difficulties experienced by some hearing-impaired listeners in understanding speech could be attributed to losses in the extraction of speech information, the integration of speech cues, or both. The present study evaluated the ability of normal-hearing and hearing-impaired listeners to integrate speech information within and across sensory modalities in order to determine the degree to which integration efficiency may be a factor in the performance of hearing-impaired listeners. Auditory-visual nonsense syllables consisting of eighteen medial consonants surrounded by the vowel [a] were processed into four nonoverlapping acoustic filter bands between 300 and 6000 Hz. A variety of one, two, three, and four filter-band combinations were presented for identification in auditory-only and auditory-visual conditions: A visual-only condition was also included. Integration efficiency was evaluated using a model of optimal integration. Results showed that normal-hearing and hearing-impaired listeners integrated information across the auditory and visual sensory modalities with a high degree of efficiency, independent of differences in auditory capabilities. However, across-frequency integration for auditory-only input was less efficient for hearing-impaired listeners. These individuals exhibited particular difficulty extracting information from the highest frequency band (4762-6000 Hz) when speech information was presented concurrently in the next lower-frequency band (1890-2381 Hz). Results suggest that integration of speech information within the auditory modality, but not across auditory and visual modalities, affects speech understanding in hearing-impaired listeners.     

Speech recognition and the Articulation Index for normal and hearing-impaired listeners

The purpose of this experiment was to determine the applicability of the Articulation Index (AI) model for characterizing the speech recognition performance of listeners with mild-to-moderate hearing loss. Performance-intensity functions were obtained from five normal-hearing listeners and 11 hearing-impaired listeners using a closed-set nonsense syllable test for two frequency responses (uniform and high-frequency emphasis). For each listener, the fitting constant Q of the nonlinear transfer function relating AI and speech recognition was estimated. Results indicated that the function mapping AI onto performance was approximately the same for normal and hearing-impaired listeners with mild-to-moderate hearing loss and high speech recognition scores. For a hearing-impaired listener with poor speech recognition ability, the AI procedure was a poor predictor of performance. The AI procedure as presently used is inadequate for predicting performance of individuals with reduced speech recognition ability and should be used conservatively in applications predicting optimal or acceptable frequency response characteristics for hearing-aid amplification systems.     

Effect of multiple speechlike maskers on binaural speech recognition in normal and impaired hearing.

Speech-reception thresholds (SRT) were measured for 17 normal-hearing and 17 hearing-impaired listeners in conditions simulating free-field situations with between one and six interfering talkers. The stimuli, speech and noise with identical long-term average spectra, were recorded with a KEMAR manikin in an anechoic room and presented to the subjects through headphones. The noise was modulated using the envelope fluctuations of the speech. Several conditions were simulated with the speaker always in front of the listener and the maskers either also in front, or positioned in a symmetrical or asymmetrical configuration around the listener. Results show that the hearing impaired have significantly poorer performance than the normal hearing in all conditions. The mean SRT differences between the groups range from 4.2-10 dB. It appears that the modulations in the masker act as an important cue for the normal-hearing listeners, who experience up to 5-dB release from masking, while being hardly beneficial for the hearing impaired listeners. The gain occurring when maskers are moved from the frontal position to positions around the listener varies from 1.5 to 8 dB for the normal hearing, and from 1 to 6.5 dB for the hearing impaired. It depends strongly on the number of maskers and their positions, but less on hearing impairment. The difference between the SRTs for binaural and best-ear listening (the "cocktail party effect") is approximately 3 dB in all conditions for both the normal-hearing and the hearing-impaired listeners.     

Discrimination of interaural temporal disparities by normal-hearing listeners and listeners with high-frequency sensorineural hearing loss

Thresholds of ongoing interaural time difference (ITD) were obtained from normal-hearing and hearing-impaired listeners who had high-frequency, sensorineural hearing loss. Several stimuli (a 500-Hz sinusoid, a narrow-band noise centered at 500 Hz, a sinusoidally amplitude-modulated 4000-Hz tone, and a narrow-band noise centered at 4000 Hz) and two criteria [equal sound-pressure level (Eq SPL) and equal sensation level (Eq SL)] for determining the level of stimuli presented to each listener were employed. The ITD thresholds and slopes of the psychometric functions were elevated for hearing-impaired listeners for the two high-frequency stimuli in comparison to: the listener's own low-frequency thresholds; and data obtained from normal-hearing listeners for stimuli presented with Eq SPL interaurally. The two groups of listeners required similar ITDs to reach threshold when stimuli were presented at Eq SLs to each ear. For low-frequency stimuli, the ITD thresholds of the hearing-impaired listener were generally slightly greater than those obtained from the normal-hearing listeners. Whether these stimuli were presented at either Eq SPL or Eq SL did not differentially affect the ITD thresholds across groups.     

The effect of hearing impairment on the identification of speech that is modulated synchronously or asynchronously across frequency

This study investigated the effect of mild-to-moderate sensorineural hearing loss on the ability to identify speech in noise for vowel-consonant-vowel tokens that were either unprocessed, amplitude modulated synchronously across frequency, or amplitude modulated asynchronously across frequency. One goal of the study was to determine whether hearing-impaired listeners have a particular deficit in the ability to integrate asynchronous spectral information in the perception of speech. Speech tokens were presented at a high, fixed sound level and the level of a speech-shaped noise was changed adaptively to estimate the masked speech identification threshold. The performance of the hearing-impaired listeners was generally worse than that of the normal-hearing listeners, but the impaired listeners showed particularly poor performance in the synchronous modulation condition. This finding suggests that integration of asynchronous spectral information does not pose a particular difficulty for hearing-impaired listeners with mild/moderate hearing losses. Results are discussed in terms of common mechanisms that might account for poor speech identification performance of hearing-impaired listeners when either the masking noise or the speech is synchronously modulated.     

Ray-tracing prediction of optimal conditions for speech in realistic classrooms

Recent papers have discussed the optimal reverberation times in classrooms for speech intelligibility, based on the assumption of a diffuse sound field. Here this question was investigated for more ‘typical’ classrooms with non-diffuse sound fields. A ray-tracing model was modified to predict speech-intelligibility metric U₅₀. It was used to predict U₅₀ in various classroom configurations for various values of the room absorption, allowing the optimal absorption (that predicting the highest U₅₀)—and the corresponding optimal reverberation time—to be identified in each case. The range of absorptions and reverberation times corresponding to high speech intelligibility were also predicted in each case. Optimal reverberation times were also predicted from the optimal surface-absorption coefficients using Sabine and Eyring versions of diffuse-field theory, and using the diffuse-field expression of Hodgson and Nosal. In order to validate the ray-tracing model, predictions were made for three classrooms with highly diffuse sound fields; these were compared to values obtained by the diffuse-field models, with good agreement. The methods were then applied to three ‘typical’ classrooms with non-diffuse fields. Optimal reverberation times increased with room volume and noise level to over 1 s. The accuracy of the Hodgson and Nosal expression varied with classroom size and noise level. The optimal average surface-absorption coefficients varied from 0.19 to 0.83 in the different classroom configurations tested. High speech intelligibility was, in general, predicted for a wide range of coefficients, but could not be obtained in a large, noisy classroom.     

Effects of degradation of intensity, time, or frequency content on speech intelligibility for normal-hearing and hearing-impaired listeners

Many hearing-impaired listeners suffer from distorted auditory processing capabilities. This study examines which aspects of auditory coding (i.e., intensity, time, or frequency) are distorted and how this affects speech perception. The distortion-sensitivity model is used: The effect of distorted auditory coding of a speech signal is simulated by an artificial distortion, and the sensitivity of speech intelligibility to this artificial distortion is compared for normal-hearing and hearing-impaired listeners. Stimuli (speech plus noise) are wavelet coded using a complex sinusoidal carrier with a Gaussian envelope (1/4 octave bandwidth). Intensity information is distorted by multiplying the modulus of each wavelet coefficient by a random factor. Temporal and spectral information are distorted by randomly shifting the wavelet positions along the temporal or spectral axis, respectively. Measured were (1) detection thresholds for each type of distortion, and (2) speech-reception thresholds for various degrees of distortion. For spectral distortion, hearing-impaired listeners showed increased detection thresholds and were also less sensitive to the distortion with respect to speech perception. For intensity and temporal distortion, this was not observed. Results indicate that a distorted coding of spectral information may be an important factor underlying reduced speech intelligibility for the hearing impaired.     

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.     

The interpretation of speech reception threshold data in normal-hearing and hearing-impaired listeners: steady-state noise

Speech-in-noise-measurements are important in clinical practice and have been the subject of research for a long time. The results of these measurements are often described in terms of the speech reception threshold (SRT) and SNR loss. Using the basic concepts that underlie several models of speech recognition in steady-state noise, the present study shows that these measures are ill-defined, most importantly because the slope of the speech recognition functions for hearing-impaired listeners always decreases with hearing loss. This slope can be determined from the slope of the normal-hearing speech recognition function when the SRT for the hearing-impaired listener is known. The SII-function (i.e., the speech intelligibility index (SII) against SNR) is important and provides insights into many potential pitfalls when interpreting SRT data. Standardized SNR loss, sSNR loss, is introduced as a universal measure of hearing loss for speech in steady-state noise. Experimental data demonstrates that, unlike the SRT or SNR loss, sSNR loss is invariant to the target point chosen, the scoring method or the type of speech material.     

Prediction of speech intelligibility for normal-hearing and cochlearly hearing-impaired listeners

The word recognition ability of 4 normal-hearing and 13 cochlearly hearing-impaired listeners was evaluated. Filtered and unfiltered speech in quiet and in noise were presented monaurally through headphones. The noise varied over listening situations with regard to spectrum, level, and temporal envelope. Articulation index theory was applied to predict the results. Two calculation methods were used, both based on the ANSI S3.5-1969 20-band method [S3.5-1969 (American National Standards Institute, New York)]. Method I was almost identical to the ANSI method. Method II included a level- and hearing-loss-dependent calculation of masking of stationary and on-off gated noise signals and of self-masking of speech. Method II provided the best prediction capability, and it is concluded that speech intelligibility of cochlearly hearing-impaired listeners may also, to a first approximation, be predicted from articulation index theory.     

Articulation index predictions of contextually dependent words

Three investigations were conducted to determine the application of the articulation index (AI) to the prediction of speech performance of hearing-impaired subjects as well as of normal-hearing listeners. Speech performance was measured in quiet and in the presence of two interfering signals for items from the Speech Perception in Noise test in which target words are either highly predictable from contextual cues in the sentence or essentially contextually neutral. As expected, transfer functions relating the AI to speech performance were different depending on the type of contextual speech material. The AI transfer function for probability-high items rises steeply, much as for sentence materials, while the function for probability-low items rises more slowly, as for monosyllabic words. Different transfer functions were also found for tests conducted in quiet or white noise rather than in a babble background. A majority of the AI predictions for ten individuals with moderate sensorineural loss fell within +/- 2 standard deviations of normal listener performance for both quiet and babble conditions.     

Prediction of optimal conditions for verbal-communication quality in eating establishments

This paper discusses the prediction of verbal-communication quality in eating establishments (EEs). EEs contain talkers and listeners who require high speech intelligibility at their tables, and high speech privacy between tables. Using catt-Acoustic, verbal-communication quality--quantified by speech transmission index (STI)--in models of three existing EEs was predicted. Talker voice-output levels were predicted using an existing empirical model accounting for the Lombard effect. With these, catt-Acoustic predicted impulse responses, speech levels and noise levels at primary and secondary listener positions, and the corresponding STIs. The untreated EEs were first modeled for various talker and listener positions, and occupancies. Then various treated configurations, involving reduced volume, increased absorption and barriers were studied to determine the effectiveness of the treatments. The results suggest that placing barriers around tables can be an effective way to achieve good verbal-communication quality. Increasing the absorption of the room surfaces or decreasing the ceiling height to control reverberation may not be effective. However, increasing the surface absorption and putting barriers around tables may achieve optimal speech conditions in EEs. Subdividing large EEs into smaller ones can also be effective.     

Effect of enhancement of spectral changes on speech intelligibility and clarity preferences for the hearing impaired

Most information in speech is carried in spectral changes over time, rather than in static spectral shape per se. A form of signal processing aimed at enhancing spectral changes over time was developed and evaluated using hearing-impaired listeners. The signal processing was based on the overlap-add method, and the degree and type of enhancement could be manipulated via four parameters. Two experiments were conducted to assess speech intelligibility and clarity preferences. Three sets of parameter values (one corresponding to a control condition), two types of masker (steady speech-spectrum noise and two-talker speech) and two signal-to-masker ratios (SMRs) were used for each masker type. Generally, the effects of the processing were small, although intelligibility was improved by about 8 percentage points relative to the control condition for one set of parameter values using the steady noise masker at -6 dB SMR. The processed signals were not preferred over those for the control condition, except for the steady noise masker at -6 dB SMR. Further work is needed to determine whether tailoring the processing to the characteristics of the individual hearing-impaired listener is beneficial.     

Effects of moderate cochlear hearing loss on the ability to benefit from temporal fine structure information in speech

Speech reception thresholds (SRTs) were measured with a competing talker background for signals processed to contain variable amounts of temporal fine structure (TFS) information, using nine normal-hearing and nine hearing-impaired subjects. Signals (speech and background talker) were bandpass filtered into channels. Channel signals for channel numbers above a "cut-off channel" (CO) were vocoded to remove TFS information, while channel signals for channel numbers of CO and below were left unprocessed. Signals from all channels were combined. As a group, hearing-impaired subjects benefited less than normal-hearing subjects from the additional TFS information that was available as CO increased. The amount of benefit varied between hearing-impaired individuals, with some showing no improvement in SRT and one showing an improvement similar to that for normal-hearing subjects. The reduced ability to take advantage of TFS information in speech may partially explain why subjects with cochlear hearing loss get less benefit from listening in a fluctuating background than normal-hearing subjects. TFS information may be important in identifying the temporal "dips" in such a background.     

The combined effects of reverberation and nonstationary noise on sentence intelligibility

Listening conditions in everyday life typically include a combination of reverberation and nonstationary background noise. It is well known that sentence intelligibility is adversely affected by these factors. To assess their combined effects, an approach is introduced which combines two methods of predicting speech intelligibility, the extended speech intelligibility index (ESII) and the speech transmission index. First, the effects of reverberation on nonstationary noise (i.e., reduction of masker modulations) and on speech modulations are evaluated separately. Subsequently, the ESII is applied to predict the speech reception threshold (SRT) in the masker with reduced modulations. To validate this approach, SRTs were measured for ten normal-hearing listeners, in various combinations of nonstationary noise and artificially created reverberation. After taking the characteristics of the speech corpus into account, results show that the approach accurately predicts SRTs in nonstationary noise and reverberation for normal-hearing listeners. Furthermore, it is shown that, when reverberation is present, the benefit from masker fluctuations may be substantially reduced.     

The influence of talker differences on vowel identification by normal-hearing and hearing-impaired listeners.

Vowel identification was tested in quiet, noise, and reverberation with 20 normal-hearing subjects and 20 hearing-impaired subjects. Stimuli were 15 English vowels spoken in a /b-t/context by six male talkers. Each talker produced five tokens of each vowel. In quiet, all stimuli were identified by two judges as the intended targets. The stimuli were degraded by reverberation or speech-spectrum noise. Vowel identification scores depended upon talker, listening condition, and subject type. The relationship between identification errors and spectral details of the vowels is discussed.     

Auditory and nonauditory factors affecting speech reception in noise by older listeners

Speech reception thresholds (SRTs) for sentences were determined in stationary and modulated background noise for two age-matched groups of normal-hearing (N = 13) and hearing-impaired listeners (N = 21). Correlations were studied between the SRT in noise and measures of auditory and nonauditory performance, after which stepwise regression analyses were performed within both groups separately. Auditory measures included the pure-tone audiogram and tests of spectral and temporal acuity. Nonauditory factors were assessed by measuring the text reception threshold (TRT), a visual analogue of the SRT, in which partially masked sentences were adaptively presented. Results indicate that, for the normal-hearing group, the variance in speech reception is mainly associated with nonauditory factors, both in stationary and in modulated noise. For the hearing-impaired group, speech reception in stationary noise is mainly related to the audiogram, even when audibility effects are accounted for. In modulated noise, both auditory (temporal acuity) and nonauditory factors (TRT) contribute to explaining interindividual differences in speech reception. Age was not a significant factor in the results. It is concluded that, under some conditions, nonauditory factors are relevant for the perception of speech in noise. Further evaluation of nonauditory factors might enable adapting the expectations from auditory rehabilitation in clinical settings.     

Temporal and spectral masking release in low- and mid-frequency regions for normal-hearing and hearing-impaired listeners

"Masking release" (MR), the improvement of speech intelligibility in modulated compared with unmodulated maskers, is typically smaller than normal for hearing-impaired listeners. The extent to which this is due to reduced audibility or to suprathreshold processing deficits is unclear. Here, the effects of audibility were controlled by using stimuli restricted to the low- (≤1.5 kHz) or mid-frequency (1-3 kHz) region for normal-hearing listeners and hearing-impaired listeners with near-normal hearing in the tested region. Previous work suggests that the latter may have suprathreshold deficits. Both spectral and temporal MR were measured. Consonant identification was measured in quiet and in the presence of unmodulated, amplitude-modulated, and spectrally modulated noise at three signal-to-noise ratios (the same ratios for the two groups). For both frequency regions, consonant identification was poorer for the hearing-impaired than for the normal-hearing listeners in all conditions. The results suggest the presence of suprathreshold deficits for the hearing-impaired listeners, despite near-normal audiometric thresholds over the tested frequency regions. However, spectral MR and temporal MR were similar for the two groups. Thus, the suprathreshold deficits for the hearing-impaired group did not lead to reduced MR.