Effective compression and noise reduction configurations for hearing protectors

The author proposed to adopt wide dynamic range compression and adaptive multichannel modulation-based noise reduction algorithms to enhance hearing protector performance. Three experiments were conducted to investigate the effects of compression and noise reduction configurations on the amount of noise reduction, speech intelligibility, and overall preferences using existing digital hearing aids. In Experiment 1, sentence materials were recorded in speech spectrum noise and white noise after being processed by eight digital hearing aids. When the hearing aids were set to 3:1 compression, the amount of noise reduction achieved was enhanced or maintained for hearing aids with parallel configurations, but reduced for hearing aids with serial configurations. In Experiments 2 and 3, 16 normal-hearing listeners' speech intelligibility and perceived sound quality were tested when they listened to speech recorded through hearing aids with parallel and serial configurations. Regardless of the configuration, the noise reduction algorithms reduced the noise level and maintained speech intelligibility in white noise. Additionally, the listeners preferred the parallel rather than the serial configuration in 3:1 conditions and the serial configuration in 1:1 rather than 3:1 compression when the noise reduction algorithms were activated. Implications for hearing protector and hearing aid design are discussed.     

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.     

Prediction of speech intelligibility in spatial noise and reverberation for normal-hearing and hearing-impaired listeners

Binaural speech intelligibility of individual listeners under realistic conditions was predicted using a model consisting of a gammatone filter bank, an independent equalization-cancellation (EC) process in each frequency band, a gammatone resynthesis, and the speech intelligibility index (SII). Hearing loss was simulated by adding uncorrelated masking noises (according to the pure-tone audiogram) to the ear channels. Speech intelligibility measurements were carried out with 8 normal-hearing and 15 hearing-impaired listeners, collecting speech reception threshold (SRT) data for three different room acoustic conditions (anechoic, office room, cafeteria hall) and eight directions of a single noise source (speech in front). Artificial EC processing errors derived from binaural masking level difference data using pure tones were incorporated into the model. Except for an adjustment of the SII-to-intelligibility mapping function, no model parameter was fitted to the SRT data of this study. The overall correlation coefficient between predicted and observed SRTs was 0.95. The dependence of the SRT of an individual listener on the noise direction and on room acoustics was predicted with a median correlation coefficient of 0.91. The effect of individual hearing impairment was predicted with a median correlation coefficient of 0.95. However, for mild hearing losses the release from masking was overestimated.     

Effects of age and mild hearing loss on speech recognition in noise

Using an adaptive strategy, the effects of mild sensorineural hearing loss and adult listeners' chronological age on speech recognition in babble were evaluated. The signal-to-babble ratio required to achieve 50% recognition was measured for three speech materials presented at soft to loud conversational speech levels. Four groups of subjects were tested: (1) normal-hearing listeners less than 44 years of age, (2) subjects less than 44 years old with mild sensorineural hearing loss and excellent speech recognition in quiet, (3) normal-hearing listeners greater than 65 with normal hearing, and (4) subjects greater than 65 years old with mild hearing loss and excellent performance in quiet. Groups 1 and 3, and groups 2 and 4 were matched on the basis of pure-tone thresholds, and thresholds for each of the three speech materials presented in quiet. In addition, groups 1 and 2 were similar in terms of mean age and age range, as were groups 3 and 4. Differences in performance in noise as a function of age were observed for both normal-hearing and hearing-impaired listeners despite equivalent performance in quiet. Subjects with mild hearing loss performed significantly worse than their normal-hearing counterparts. These results and their implications are discussed.     

Intelligibility ratings of continuous discourse: application to hearing aid selection

Twelve normal-hearing subjects rated the intelligibility of 35-s, hearing-aid-processed continuous discourse (CD) passages. Three talkers (two male, one female), four hearing aids, and two signal-to-babble (S/B) ratios were used in a completely crossed design. Research questions concerned: (1) ability of listeners to rate intelligibility, (2) sensitivity of hearing aid rankings were based on intelligibility ratings for three CD passages per instrument, and (3) dependence of hearing aid rankings on (a) S/B ratio, and (b) talker characteristics. Results were: (1) listeners were able to rate intelligibility, (2) rankings based on intelligibility ratings of three CD passages per hearing aid were capable of identifying two superior instruments within a group of four hearing aids that were similar in frequency/gain function, (3) listening in a more difficult S/B ratio substantially decreased the sensitivity of the hearing aid rankings for the female talker but had only minor effects on the rankings for the male talkers, and (4) hearing aid intelligibility rankings were found to be different for different talkers. Applications to hearing aid selection are discussed.     

An evaluation of three adaptive hearing aid selection strategies

Paired-comparison judgments of intelligibility of speech in noise were obtained from eight hearing-impaired subjects on a large number of hearing aids simulated by a digital master hearing aid. The hearing aids which comprised a 5 X 5 matrix differed systematically in the amount of low-frequency and high-frequency gain provided. A comparison of three adaptive strategies for determining optimum hearing aid frequency-gain characteristics (an iterative round robin, a double elimination tournament, and a modified simplex procedure) revealed convergence on the same or similar hearing aids for most subjects. Analysis revealed that subjects for whom all three procedures converged on the same hearing aid showed a single pronounced peak in the response surface, while a broader peak was evident for the subjects for whom the three procedures identified similar hearing aids. The modified simplex procedure was found to be most efficient and the iterative round robin least efficient.     

Speech intelligibility in cochlear implant simulations: Effects of carrier type, interfering noise, and subject experience

Channel vocoders using either tone or band-limited noise carriers have been used in experiments to simulate cochlear implant processing in normal-hearing listeners. Previous results from these experiments have suggested that the two vocoder types produce speech of nearly equal intelligibility in quiet conditions. The purpose of this study was to further compare the performance of tone and noise-band vocoders in both quiet and noisy listening conditions. In each of four experiments, normal-hearing subjects were better able to identify tone-vocoded sentences and vowel-consonant-vowel syllables than noise-vocoded sentences and syllables, both in quiet and in the presence of either speech-spectrum noise or two-talker babble. An analysis of consonant confusions for listening in both quiet and speech-spectrum noise revealed significantly different error patterns that were related to each vocoder's ability to produce tone or noise output that accurately reflected the consonant's manner of articulation. Subject experience was also shown to influence intelligibility. Simulations using a computational model of modulation detection suggest that the noise vocoder's disadvantage is in part due to the intrinsic temporal fluctuations of its carriers, which can interfere with temporal fluctuations that convey speech recognition cues.     

Effects of low-pass filtering on intelligibility of periodically interrupted speech

The combined effect of low-pass filtering (cut-off frequencies between 500 and 3000 Hz) and periodic interruptions (1.5 and 10 Hz) on speech intelligibility was investigated. When combined, intelligibility was lower than each manipulation alone, even in some conditions where there was no effect from a single manipulation (such as the fast interruption rate of 10 Hz). By using young normal-hearing listeners, potential suprathreshold deficits and aging effects that may occur due to hearing impairment were eliminated. Thus, the results imply that reduced audibility of high-frequency speech components may partially explain the reduced intelligibility of interrupted speech in hearing impaired persons.     

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.     

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.     

The acoustic and perceptual effects of two noise-suppression algorithms

Internal noise generated by hearing-aid circuits can be audible and objectionable to aid users, and may lead to the rejection of hearing aids. Two expansion algorithms were developed to suppress internal noise below a threshold level. The multiple-channel algorithm's expansion thresholds followed the 55-dB SPL long-term average speech spectrum, while the single-channel algorithm suppressed sounds below 45 dBA. With the recommended settings in static conditions, the single-channel algorithm provided lower noise levels, which were perceived as quieter by most normal-hearing participants. However, in dynamic conditions "pumping" noises were more noticeable with the single-channel algorithm. For impaired-hearing listeners fitted with the ADRO amplification strategy, both algorithms maintained speech understanding for words in sentences presented at 55 dB SPL in quiet (99.3% correct). Mean sentence reception thresholds in quiet were 39.4, 40.7, and 41.8 dB SPL without noise suppression, and with the single- and multiple-channel algorithms, respectively. The increase in the sentence reception threshold was statistically significant for the multiple-channel algorithm, but not the single-channel algorithm. Thus, both algorithms suppressed noise without affecting the intelligibility of speech presented at 55 dB SPL, with the single-channel algorithm providing marginally greater noise suppression in static conditions, and the multiple-channel algorithm avoiding pumping noises.     

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.     

Monosyllabic word recognition at higher-than-normal speech and noise levels

The effects of intensity on monosyllabic word recognition were studied in adults with normal hearing and mild-to-moderate sensorineural hearing loss. The stimuli were bandlimited NU#6 word lists presented in quiet and talker-spectrum-matched noise. Speech levels ranged from 64 to 99 dB SPL and S/N ratios from 28 to -4 dB. In quiet, the performance of normal-hearing subjects remained essentially constant in noise, at a fixed S/N ratio, it decreased as a linear function of speech level. Hearing-impaired subjects performed like normal-hearing subjects tested in noise when the data were corrected for the effects of audibility loss. From these and other results, it was concluded that: (1) speech intelligibility in noise decreases when speech levels exceed 69 dB SPL and the S/N ratio remains constant; (2) the effects of speech and noise level are synergistic; (3) the deterioration in intelligibility can be modeled as a relative increase in the effective masking level; (4) normal-hearing and hearing-impaired subjects are affected similarly by increased signal level when differences in speech audibility are considered; (5) the negative effects of increasing speech and noise levels on speech recognition are similar for all adult subjects, at least up to 80 years; and (6) the effective dynamic range of speech may be larger than the commonly assumed value of 30 dB.     

Contribution of consonant versus vowel information to sentence intelligibility for young normal-hearing and elderly hearing-impaired listeners

The purpose of this study was to examine the contribution of information provided by vowels versus consonants to sentence intelligibility in young normal-hearing (YNH) and typical elderly hearing-impaired (EHI) listeners. Sentences were presented in three conditions, unaltered or with either the vowels or the consonants replaced with speech shaped noise. Sentences from male and female talkers in the TIMIT database were selected. Baseline performance was established at a 70 dB SPL level using YNH listeners. Subsequently EHI and YNH participants listened at 95 dB SPL. Participants listened to each sentence twice and were asked to repeat the entire sentence after each presentation. Words were scored correct if identified exactly. Average performance for unaltered sentences was greater than 94%. Overall, EHI listeners performed more poorly than YNH listeners. However, vowel-only sentences were always significantly more intelligible than consonant-only sentences, usually by a ratio of 2:1 across groups. In contrast to written English or words spoken in isolation, these results demonstrated that for spoken sentences, vowels carry more information about sentence intelligibility than consonants for both young normal-hearing and elderly hearing-impaired listeners.     

Consonant reception in noise by listeners with mild and moderate sensorineural hearing impairment

The goal of this study was to determine the extent to which the difficulty experienced by impaired listeners in understanding noisy speech can be explained on the basis of elevated tone-detection thresholds. Twenty-one impaired ears of 15 subjects, spanning a variety of audiometric configurations with average hearing losses to 75 dB, were tested for reception of consonants in a speech-spectrum noise. Speech level, noise level, and frequency-gain characteristic were varied to generate a range of listening conditions. Results for impaired listeners were compared to those of normal-hearing listeners tested under the same conditions with extra noise added to approximate the impaired listeners' detection thresholds. Results for impaired and normal listeners were also compared on the basis of articulation indices. Consonant recognition by this sample of impaired listeners was generally comparable to that of normal-hearing listeners with similar threshold shifts listening under the same conditions. When listening conditions were equated for articulation index, there was no clear dependence of consonant recognition on average hearing loss. Assuming that the primary consequence of the threshold simulation in normals is loss of audibility (as opposed to suprathreshold discrimination or resolution deficits), it is concluded that the primary source of difficulty in listening in noise for listeners with moderate or milder hearing impairments, aside from the noise itself, is the loss of audibility.     

Gain-induced speech distortions and the absence of intelligibility benefit with existing noise-reduction algorithms

Most noise-reduction algorithms used in hearing aids apply a gain to the noisy envelopes to reduce noise interference. The present study assesses the impact of two types of speech distortion introduced by noise-suppressive gain functions: amplification distortion occurring when the amplitude of the target signal is over-estimated, and attenuation distortion occurring when the target amplitude is under-estimated. Sentences corrupted by steady noise and competing talker were processed through a noise-reduction algorithm and synthesized to contain either amplification distortion, attenuation distortion or both. The attenuation distortion was found to have a minimal effect on speech intelligibility. In fact, substantial improvements (>80 percentage points) in intelligibility, relative to noise-corrupted speech, were obtained when the processed sentences contained only attenuation distortion. When the amplification distortion was limited to be smaller than 6 dB, performance was nearly unaffected in the steady-noise conditions, but was severely degraded in the competing-talker conditions. Overall, the present data suggest that one reason that existing algorithms do not improve speech intelligibility is because they allow amplification distortions in excess of 6 dB. These distortions are shown in this study to be always associated with masker-dominated envelopes and should thus be eliminated.     

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.     

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.     

Compression and expansion of the temporal envelope: evaluation of speech intelligibility and sound quality.

Sensorineural hearing loss is accompanied by loudness recruitment, a steeper-than-normal rise of perceived loudness with presentation level. To compensate for this abnormality, amplitude compression is often applied (e.g., in a hearing aid). Alternatively, since speech intelligibility has been modeled as the perception of fast energy fluctuations, enlarging these (by means of expansion) may improve speech intelligibility. Still, even if these signal-processing techniques prove useful in terms of speech intelligibility, practical application might be hindered by unacceptably low sound quality. Therefore, both speech intelligibility and sound quality were evaluated for syllabic compression and expansion of the temporal envelope. Speech intelligibility was evaluated with an adaptive procedure, based on short everyday sentences either in noise or with a competing speaker. Sound quality was measured by means of a rating-scale procedure, for both speech and music. In a systematic setup, both the ratio of compression or expansion and the number of independent processing bands were varied. Individual hearing thresholds were compensated for by a listener-specific filter and amplification. Both listeners with normal hearing and listeners with sensorineural hearing impairment participated as paid volunteers. The results show that, on average, both compression and expansion fail to show better speech intelligibility or sound quality than linear amplification.     

Comparative intelligibility investigation of single-channel noise-reduction algorithms for Chinese, Japanese, and English

A large number of single-channel noise-reduction algorithms have been proposed based largely on mathematical principles. Most of these algorithms, however, have been evaluated with English speech. Given the different perceptual cues used by native listeners of different languages including tonal languages, it is of interest to examine whether there are any language effects when the same noise-reduction algorithm is used to process noisy speech in different languages. A comparative evaluation and investigation is taken in this study of various single-channel noise-reduction algorithms applied to noisy speech taken from three languages: Chinese, Japanese, and English. Clean speech signals (Chinese words and Japanese words) were first corrupted by three types of noise at two signal-to-noise ratios and then processed by five single-channel noise-reduction algorithms. The processed signals were finally presented to normal-hearing listeners for recognition. Intelligibility evaluation showed that the majority of noise-reduction algorithms did not improve speech intelligibility. Consistent with a previous study with the English language, the Wiener filtering algorithm produced small, but statistically significant, improvements in intelligibility for car and white noise conditions. Significant differences between the performances of noise-reduction algorithms across the three languages were observed.