Acceptable range of speech level for both young and aged listeners in reverberant and quiet sound fields

The speech level of verbal information in public spaces should be determined to make it acceptable to as many listeners as possible, while simultaneously maintaining maximum intelligibility and considering the variation in the hearing levels of listeners. In the present study, the universally acceptable range of speech level in reverberant and quiet sound fields for both young listeners with normal hearing and aged listeners with hearing loss due to aging was investigated. Word intelligibility scores and listening difficulty ratings as a function of speech level were obtained by listening tests. The results of the listening tests clarified that (1) the universally acceptable ranges of speech level are from 60 to 70 dBA, from 56 to 61 dBA, from 52 to 67 dBA and from 58 to 63 dBA for the test sound fields with the reverberation times of 0.0, 0.5, 1.0 and 2.0 s, respectively, and (2) there is a speech level that falls within all of the universally acceptable ranges of speech level obtained in the present study; that speech level is around 60 dBA.     

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.     

Optimum speech level to minimize listening difficulty in public spaces

For ideal speech communication in public spaces, it is important to determine the optimum speech level for various background noise levels. However, speech intelligibility scores, which is conventionally used as the subjective listening test to measure the quality of speech communication, is near perfect in most everyday situations. For this reason, it is proposed to determine optimum speech levels for speech communication in public spaces by using listening difficulty ratings. Two kinds of listening test were carried out in this work. The results of the tests and our previous work [M. Morimoto, H. Sato, and M. Kobayashi, J. Acoust. Soc. Am. 116, 1607-1613 (2004)] are jointly discussed for suggesting the relation between the optimum speech level and background noise level. The results demonstrate that: (1) optimum speech level is constant when background noise level is lower than 40 dBA, (2) optimum speech level appears to be the level, which maintains around 15 dBA of SN ratio when the background noise level is more than 40 dBA, and (3) listening difficulty increases as speech level increases under the condition where SN ratio is good enough to keep intelligibility near perfect.     

Speech intelligibility of normal listeners and persons with impaired hearing in traffic noise

Speech intelligibility (PB words) in traffic-like noise was investigated in a laboratory situation simulating three common listening situations, indoors at 1 and 4 m and outdoors at 1 m. The maximum noise levels still permitting 75% intelligibility of PB words in these three listening situations were also defined. A total of 269 persons were examined. Forty-six had normal hearing, 90 a presbycusis-type hearing loss, 95 a noise-induced hearing loss and 38 a conductive hearing loss. In the indoor situation the majority of the groups with impaired hearing retained good speech intelligibility in 40 dB(A) masking noise. Lowering the noise level to less than 40 dB(A) resulted in a minor, usually insignificant, improvement in speech intelligibility. Listeners with normal hearing maintained good speech intelligibility in the outdoor listening situation at noise levels up to 60 dB(A), without lip-reading (i.e., using non-auditory information). For groups with impaired hearing due to age and/or noise, representing 8% of the population in Sweden, the noise level outdoors had to be lowered to less than 50 dB(A), in order to achieve good speech intelligibility at 1 m without lip-reading.     

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.     

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.     

Quantifying the intelligibility of speech in noise for non-native listeners

When listening to languages learned at a later age, speech intelligibility is generally lower than when listening to one's native language. The main purpose of this study is to quantify speech intelligibility in noise for specific populations of non-native listeners, only broadly addressing the underlying perceptual and linguistic processing. An easy method is sought to extend these quantitative findings to other listener populations. Dutch subjects listening to Germans and English speech, ranging from reasonable to excellent proficiency in these languages, were found to require a 1-7 dB better speech-to-noise ratio to obtain 50% sentence intelligibility than native listeners. Also, the psychometric function for sentence recognition in noise was found to be shallower for non-native than for native listeners (worst-case slope around the 50% point of 7.5%/dB, compared to 12.6%/dB for native listeners). Differences between native and non-native speech intelligibility are largely predicted by linguistic entropy estimates as derived from a letter guessing task. Less effective use of context effects (especially semantic redundancy) explains the reduced speech intelligibility for non-native listeners. While measuring speech intelligibility for many different populations of listeners (languages, linguistic experience) may be prohibitively time consuming, obtaining predictions of non-native intelligibility from linguistic entropy may help to extend the results of this study to other listener populations.     

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

The speech understanding of persons with "flat" hearing loss (HI) was compared to a normal-hearing (NH) control group to examine how hearing loss affects the contribution of speech information in various frequency regions. Speech understanding in noise was assessed at multiple low- and high-pass filter cutoff frequencies. Noise levels were chosen to ensure that the noise, rather than quiet thresholds, determined audibility. The performance of HI subjects was compared to a NH group listening at the same signal-to-noise ratio and a comparable presentation level. Although absolute speech scores for the HI group were reduced, performance improvements as the speech and noise bandwidth increased were comparable between groups. These data suggest that the presence of hearing loss results in a uniform, rather than frequency-specific, deficit in the contribution of speech information. Measures of auditory thresholds in noise and speech intelligibility index (SII) calculations were also performed. These data suggest that differences in performance between the HI and NH groups are due primarily to audibility differences between groups. Measures of auditory thresholds in noise showed the "effective masking spectrum" of the noise was greater for the HI than the NH subjects.     

Comparison of amplification systems in an auditorium

Intelligibility of speech at two positions in a large auditorium was compared for the public address system (PA) and two assistive listening systems: Frequency modulation of radio frequencies (FM) and modulation of infrared light waves (IR). Listening groups were: normal-hearing adults, hearing-impaired, hearing aid users, elderly, and non-native. Word-identification scores were obtained with the Modified Rhyme Tests. Analysis of variance indicated that the main effects of systems, groups, and listening position were significant. Also significant were the two-way interactions. For all groups, the assistive listening systems provided better scores than the PA system. The difference between the two systems was statistically significant, but very small. It can be concluded that both listening systems provide improved speech intelligibility for various types of listeners.     

Traffic noise annoyance and speech intelligibility in persons with normal and persons with impaired hearing

Annoyance ratings in speech intelligibility tests at 45 dB(A) and 55 dB(A) traffic noise were investigated in a laboratory study. Subjects were chosen according to their hearing acuity to be representative of 70-year-old men and women, and of noise-induced hearing losses typical for a great number of industrial workers. These groups were compared with normal hearing subjects of the same sex and, when possible, the same age. The subjects rated their annoyance on an open 100 mm scale. Significant correlations were found between annoyance expressed in millimetres and speech intelligibility in percent when all subjects were taken as one sample. Speech intelligibility was also calculated from physical measurements of speech and noise by using the articulation index method. Observed and calculated speech intelligibility scores are compared and discussed. Also treated is the estimation of annoyance by traffic noise at moderate noise levels via speech intelligibility scores.     

Recognition of accented English in quiet and noise by younger and older listeners

This study investigated the effects of age and hearing loss on perception of accented speech presented in quiet and noise. The relative importance of alterations in phonetic segments vs. temporal patterns in a carrier phrase with accented speech also was examined. English sentences recorded by a native English speaker and a native Spanish speaker, together with hybrid sentences that varied the native language of the speaker of the carrier phrase and the final target word of the sentence were presented to younger and older listeners with normal hearing and older listeners with hearing loss in quiet and noise. Effects of age and hearing loss were observed in both listening environments, but varied with speaker accent. All groups exhibited lower recognition performance for the final target word spoken by the accented speaker compared to that spoken by the native speaker, indicating that alterations in segmental cues due to accent play a prominent role in intelligibility. Effects of the carrier phrase were minimal. The findings indicate that recognition of accented speech, especially in noise, is a particularly challenging communication task for older people.     

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.     

密闭环境条件下人的听觉功能变化

本文研究了10名受试者在密闭环境条件下听觉功能的变化。环境因素包括噪声、温度、湿度、脑力负荷和某些微量气体成分。受试者的听阈偏移在实验期15—30天后即趋于稳定,在实验终止后15天恢复到原有水平。最大听阈偏移为20—40分贝,主要频率为1000—4000赫,与噪声频谱一致。在实验期90天最大清晰度值下降10％,可懂度阈增高7分贝。音调辨别阈在60—90天持续升高。刺激后疲劳测定表明,尽管实验后15天听阈已恢复,但听觉系统的疲劳状态仍然存在。文中讨论了听觉分析器外周和中枢部分的变化与噪声等环境因素的关系。     

Speech enhancement based on the discrete Gabor transform and multi-notch adaptive digital filters

This paper presents a new method to speech enhancement based on time-frequency analysis and adaptive digital filtering. The proposed method for dual-channel speech enhancement was developed by tracking frequencies of corrupting signal by the discrete Gabor transform (DGT) and implementing multi-notch adaptive digital filter (MNADF) at those frequencies. Since no a priori knowledge of the noise source statistics is required this method differs from traditional speech enhancement methods. Specifically, the proposed method was applied to the case where speech quality and intelligibility deteriorate in the presence of background noise. Speech coders and automatic speech recognition (ASR) systems are designed to act on clean speech signals. Therefore, corrupted speech signals by the noise must be enhanced before their processing. The method uses a primary input containing the corrupted speech signal while a reference input containing the noise only. In this paper, we designed MNADF instead of single-notch adaptive digital filter and used DGT to track frequencies of corrupting signal because fast filtering process and fast measure of the time-dependent noise frequency are of great importance in speech enhancement process. Therefore, MNADF was implemented to take advantage of fast filtering process. Different types of noises from Noisex-92 database were used to degrade real speech signals. Objective measures, the study of the speech spectrograms and global signal-to-noise ratio (SNR), segmental SNR (segSNR), Itakura-Saito distance measure as well as subjective listing test demonstrated consistently superior enhancement performance of the proposed method over traditional speech enhancement method such as spectral subtraction. Combining MNADF and DGT, excellent speech enhancement was obtained.     

Intelligibility of reverberant noisy speech with ideal binary masking

For a mixture of target speech and noise in anechoic conditions, the ideal binary mask is defined as follows: It selects the time-frequency units where target energy exceeds noise energy by a certain local threshold and cancels the other units. In this study, the definition of the ideal binary mask is extended to reverberant conditions. Given the division between early and late reflections in terms of speech intelligibility, three ideal binary masks can be defined: an ideal binary mask that uses the direct path of the target as the desired signal, an ideal binary mask that uses the direct path and early reflections of the target as the desired signal, and an ideal binary mask that uses the reverberant target as the desired signal. The effects of these ideal binary mask definitions on speech intelligibility are compared across two types of interference: speech shaped noise and concurrent female speech. As suggested by psychoacoustical studies, the ideal binary mask based on the direct path and early reflections of target speech outperforms the other masks as reverberation time increases and produces substantial reductions in terms of speech reception threshold for normal hearing listeners.     

Potential benefits of adaptive frequency-gain characteristics for speech reception in noise.

Some current single-microphone hearing aids employ techniques for adaptively varying the frequency-gain characteristics in an attempt to improve speech reception in noise. The potential benefit of this strategy depends on the spectral spread of masking and the degree to which it can be reduced by changing the frequency-gain characteristic. In this study these benefits were examined for subjects with normal hearing under static listening conditions. In the unprocessed condition, subjects were presented with nonsense syllables in an octave-band noise centered on 0.5, 1, or 2 kHz. The frequency-gain characteristic was then modified with the goal of reducing the intensity of the frequency region containing the octave-band noise. This processing resulted in increases as large as 60 percentage points in consonant-correct scores with the low- and mid-frequency octave noise bands, and a small increase with the high-frequency noise. Masking patterns produced by the octave noises were also measured and were related to the intelligibility results via an analysis based on Articulation Theory. The Articulation Index was also used to compare the effectiveness of three adaptive rules. A simple multiband volume control is expected to provide much of the benefit of more sophisticated systems without the need for separate estimation of input speech and noise spectra.     

Enhancing intelligibility of narrowband speech with out-of-band noise: evidence for lateral suppression at high-normal intensity

Previous studies have shown that the intelligibility of filtered speech can be enhanced by filling stopbands with noise. The present study found that this enhancement occurred only when speech intensity was sufficiently high to degrade performance. Intelligibility decreased by about 15% when narrowband speech was increased from 45 to 65 dBA (corresponding to broadband speech levels of about 60 and 80 dBA), and decreased by 20% at a level of 75 dBA. However, when flanking bands of low-pass and high-pass filtered white noise were added at spectrum levels of -40 to -20 dB relative to the speech, intelligibility of the 75-dBA speech band increased by about 13%. Additional findings confirm that this enhancement of intelligibility depends upon out-of-band stimulation, in agreement with theories proposing that lateral suppressive interactions extend the dynamic range of intensity coding by counteracting effects of auditory-nerve firing-rate saturation at high signal levels.     

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.     

Speech reception in quiet and in noisy conditions by individuals with noise-induced hearing loss in relation to their tone audiogram.

Tone thresholds and speech-reception thresholds were measured in 200 individuals (400 ears) with noise-induced hearing loss. The speech-reception thresholds were measured in a quiet condition and in noise with a speech spectrum at levels of 35, 50, 65, and 80 dBA. The tone audiograms could be described by three principal components: hearing loss in the regions above 3 kHz, from 1 to 3 kHz and below 1 kHz; the speech thresholds could be described by two components: speech reception in quiet and speech reception in noise at 50-80 dBA. Hearing loss above 1 kHz was related to speech reception in noise; hearing loss at and below 1 kHz to speech reception in quiet. The correlation between the speech thresholds in quiet and in noise was only R = 0.45. An adequate predictor of the speech threshold in noise, the primary factor in the hearing handicap, was the pure-tone average at 2 and 4 kHz (PTA2,4, R = 0.72). The minimum value of the prediction error for any tone-audiometric predictor of this speech threshold was 1.2 dB (standard deviation). The prediction could not be improved by taking into account the critical ratio for low-frequency noise nor by its upward spread of masking. The prediction error is due to measurement error and to a factor common to both ears. The latter factor is ascribed to cognitive skill in speech reception. Hearing loss above 10 to 15 dB HL (hearing level) already shows an effect on the speech threshold in noise, a noticeable handicap is found at PTA2,4 = 30 dB HL.     

Speech intelligibility and localization in a multi-source environment.

Natural environments typically contain sound sources other than the source of interest that may interfere with the ability of listeners to extract information about the primary source. Studies of speech intelligibility and localization by normal-hearing listeners in the presence of competing speech are reported on in this work. One, two or three competing sentences [IEEE Trans. Audio Electroacoust. 17(3), 225-246 (1969)] were presented from various locations in the horizontal plane in several spatial configurations relative to a target sentence. Target and competing sentences were spoken by the same male talker and at the same level. All experiments were conducted both in an actual sound field and in a virtual sound field. In the virtual sound field, both binaural and monaural conditions were tested. In the speech intelligibility experiment, there were significant improvements in performance when the target and competing sentences were spatially separated. Performance was similar in the actual sound-field and virtual sound-field binaural listening conditions for speech intelligibility. Although most of these improvements are evident monaurally when using the better ear, binaural listening was necessary for large improvements in some situations. In the localization experiment, target source identification was measured in a seven-alternative absolute identification paradigm with the same competing sentence configurations as for the speech study. Performance in the localization experiment was significantly better in the actual sound-field than in the virtual sound-field binaural listening conditions. Under binaural conditions, localization performance was very good, even in the presence of three competing sentences. Under monaural conditions, performance was much worse. For the localization experiment, there was no significant effect of the number or configuration of the competing sentences tested. For these experiments, the performance in the speech intelligibility experiment was not limited by localization ability.