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.     

Comparison of different forms of compression using wearable digital hearing aids

Four different compression algorithms were implemented in wearable digital hearing aids: (1) The slow-acting dual-front-end automatic gain control (AGC) system [B. C. J. Moore, B. R. Glasberg, and M. A. Stone, Br. J. Audiol. 25, 171-182 (1991)], combined with appropriate frequency response equalization, with a compression threshold of 63 dB sound pressure level (SPL) and with a compression ratio of 30 (DUAL-HI); (2) The dual-front-end AGC system combined with appropriate frequency response equalization, with a compression threshold of 55 dB SPL and with a compression ratio of 3 (DUAL-LO). This was intended to give some impression of the levels of sounds in the environment; (3) Fast-acting full dynamic range compression in four channels (FULL-4). The compression was designed to minimize envelope distortion due to overshoots and undershoots; (4) A combination of (2) and (3) above, where each applied less compression than when used alone (DUAL-4). Initial fitting was partly based on the concept of giving a flat specific-loudness pattern for a 65-dB SPL speech-shaped noise input, and this was followed by fine tuning using an adaptive procedure with speech stimuli. Eight subjects with moderate to severe cochlear hearing loss were tested in a counter-balanced design. Subjects had at least 2 weeks experience with each system in everyday life before evaluation using the Abbreviated Profile of Hearing Aid Benefit (APHAB) test and measures of speech intelligibility in quiet (AB word lists at 50 and 80 dB SPL) and noise (adoptive sentence lists in speech-shaped noise, or that same noise amplitude modulated with the envelope of speech from a single talker). The APHAB scores did not indicate clear differences between the four systems. Scores for the AB words in quiet were high for all four systems at both 50 and 80 dB SPL. The speech-to-noise ratios required for 50% intelligibility were low (indicating good performance) and similar for all the systems, but there was a slight trend for better performance in modulated noise with the DUAL-4 system than with the other systems. A subsequent trial where three subjects directly compared each of the four systems in their everyday lives indicated a slight preference for the DUAL-LO system. Overall, the results suggest that it is not necessary to compress fast modulations of the input signal.     

Evaluation of array-processing algorithms for a headband hearing aid

Several array-processing algorithms were implemented and evaluated with experienced hearing-aid users. The array consisted of four directional microphones mounted broadside on a headband worn on the top of the listener's head. The algorithms included two adaptive array-processing algorithms, one fixed array-processing algorithm, and a reference condition consisting of binaural directional microphones. The algorithms were evaluated under conditions with both one and three independent noise sources. Performance metrics included quantitative speech reception thresholds and qualitative subject preference ratings for ease-of-listening measured using a paired-comparison procedure. On average, the fixed algorithm improved speech reception thresholds by 2 dB, while the adaptive algorithms provided 7-9-dB improvement over the reference condition. Subjects judging ease-of-listening generally preferred all array-processing algorithms over the reference condition. The results suggest that these adaptive algorithms should be evaluated further in more realistic acoustic environments.     

Detection of tones in noise and the "severe departure" from Weber's law

Thresholds were measured for the detection of 20-ms sinusoids, with frequencies 500, 4000, or 6500 Hz, presented in bursts of bandpass noise of the same duration and centered around the signal frequency. A range of noise levels from 35 to 80 dB SPL was used. Noise at different center frequencies was equated in terms of the total noise power in an assumed auditory filter centered on the signal frequency. Thresholds were expressed as the signal levels, relative to these noise levels, necessary for subjects to achieve 71% correct. For 500-Hz signals, thresholds were about 5 dB regardless of noise level. For 6500-Hz signals, thresholds reached a maximum of 14 dB at intermediate noise levels of 55-65 dB SPL. For 4000-Hz signals, a maximum threshold of 10 dB was observed for noise levels of 45-55 dB SPL. When the bandpass noises were presented continuously, however, thresholds for 6500-Hz, 20-ms signals remained low (about 1 dB) and constant across level. These results are similar to those obtained for the intensity discrimination of brief tones in bandstop noise [R. P. Carlyon and B. C. J. Moore, J. Acoust. Soc. Am. 76, 1369-1376 (1984); R. P. Carlyon and B. C. J. Moore, J. Acoust. Soc. Am. 79, 453-460 (1986)].     

Companding to improve cochlear-implant speech recognition in speech-shaped noise

Nonlinear sensory and neural processing mechanisms have been exploited to enhance spectral contrast for improvement of speech understanding in noise. The "companding" algorithm employs both two-tone suppression and adaptive gain mechanisms to achieve spectral enhancement. This study implemented a 50-channel companding strategy and evaluated its efficiency as a front-end noise suppression technique in cochlear implants. The key parameters were identified and evaluated to optimize the companding performance. Both normal-hearing (NH) listeners and cochlear-implant (CI) users performed phoneme and sentence recognition tests in quiet and in steady-state speech-shaped noise. Data from the NH listeners showed that for noise conditions, the implemented strategy improved vowel perception but not consonant and sentence perception. However, the CI users showed significant improvements in both phoneme and sentence perception in noise. Maximum average improvement for vowel recognition was 21.3 percentage points (p<0.05) at 0 dB signal-to-noise ratio (SNR), followed by 17.7 percentage points (p<0.05) at 5 dB SNR for sentence recognition and 12.1 percentage points (p<0.05) at 5 dB SNR for consonant recognition. While the observed results could be attributed to the enhanced spectral contrast, it is likely that the corresponding temporal changes caused by companding also played a significant role and should be addressed by future studies.     

听觉频域掩蔽效应的自适应β阶贝叶斯感知估计语音增强算法

为了在噪声抑制和语音失真中之间寻找最佳平衡,提出了一种听觉频域掩蔽效应的自适应β阶贝叶斯感知估计语音增强算法,以期提高语音增强的综合性能。算法利用了人耳的听觉掩蔽效应,根据计算得到的频域掩蔽阈自适应调整β阶贝叶斯感知估计语音增强算法中的β值,从而仅将噪声抑制在掩蔽阈之下,保留较多的语音信息,降低语音失真。并分别用客观和主观评价方式,对所提出的算法的性能进行了评估,并与原来基于信噪比的自适应β阶贝叶斯感知估计语音增强算法进行了比较。结果表明,频域掩蔽的β阶贝叶斯感知估计方法的综合客观评价结果在信噪比为-10 dB至5 dB之间时均高于基于信噪比的自适应β阶贝叶斯感知估计语音增强算法。主观评价结果也表明频域掩蔽的β阶贝叶斯感知估计方法能在尽量保留语音信息的同时,较好的抑制背景噪声。      

Effects of noise on speech production: acoustic and perceptual analyses

Acoustical analyses were carried out on a set of utterances produced by two male speakers talking in quiet and in 80, 90, and 100 dB SPL of masking noise. In addition to replicating previous studies demonstrating increases in amplitude, duration, and vocal pitch while talking in noise, these analyses also found reliable differences in the formant frequencies and short-term spectra of vowels. Perceptual experiments were also conducted to assess the intelligibility of utterances produced in quiet and in noise when they were presented at equal S/N ratios for identification. In each experiment, utterances originally produced in noise were found to be more intelligible than utterances produced in the quiet. The results of the acoustic analyses showed clear and consistent differences in the acoustic-phonetic characteristics of speech produced in quiet versus noisy environments. Moreover, these accounts differences produced reliable effects on intelligibility. The findings are discussed in terms of: (1) the nature of the acoustic changes that taken place when speakers produce speech under adverse conditions such as noise, psychological stress, or high cognitive load: (2) the role of training and feedback in controlling and modifying a talker's speech to improve performance of current speech recognizers; and (3) the development of robust algorithms for recognition of speech in noise.     

Detection and intensity discrimination of a sinusoid

Intensity discrimination thresholds were measured for gated 100-ms, 1000-Hz tones. Discrimination thresholds were measured at several intensities near absolute threshold as well as at 30, 60, and 90 dB SPL. Psychometric functions were obtained for several of these discrimination conditions, and for detection of the signal in quiet. The results showed that Weber's law is approximately valid for standards as low as 0 dB SL. Small amounts of negative masking were observed even when the data were expressed in terms of increment energy. The psychometric functions for the discrimination conditions had a similar form and were shallower than the psychometric function for the detection of a signal in quiet. A similar set of conditions was run in the presence of a continuous, broadband noise. The results were generally in agreement with those obtained in quiet, but slight differences suggested that the variability which limits performance in the two conditions is different. The results are discussed in terms of the effects of nonlinear transduction, the effects of uncertainty, and contrast mechanisms as proposed by Laming [Sensory Analysis (Academic, London, 1986)].     

The effects of compression ratio, signal-to-noise ratio, and level on speech recognition in normal-hearing listeners.

Previous research has demonstrated reduced speech recognition when speech is presented at higher-than-normal levels (e.g., above conversational speech levels), particularly in the presence of speech-shaped background noise. Persons with hearing loss frequently listen to speech-in-noise at these levels through hearing aids, which incorporate multiple-channel, wide dynamic range compression. This study examined the interactive effects of signal-to-noise ratio (SNR), speech presentation level, and compression ratio on consonant recognition in noise. Nine subjects with normal hearing identified CV and VC nonsense syllables in a speech-shaped noise at two SNRs (0 and +6 dB), three presentation levels (65, 80, and 95 dB SPL) and four compression ratios (1:1, 2:1, 4:1, and 6:1). Stimuli were processed through a simulated three-channel, fast-acting, wide dynamic range compression hearing aid. Consonant recognition performance decreased as compression ratio increased and presentation level increased. Interaction effects were noted between SNR and compression ratio, as well as between presentation level and compression ratio. Performance decrements due to increases in compression ratio were larger at the better (+6 dB) SNR and at the lowest (65 dB SPL) presentation level. At higher levels (95 dB SPL), such as those experienced by persons with hearing loss, increasing compression ratio did not significantly affect speech intelligibility.     

Upward spread of masking, hearing loss, and speech recognition in young and elderly listeners

Upward spreading of masking, measured in terms of absolute masked threshold, is greater in hearing-impaired listeners than in listeners with normal hearing. The purpose of this study was to make further observations on upward-masked thresholds and speech recognition in noise in elderly listeners. Two age groups were used: One group consisted of listeners who were more than 60 years old, and the second group consisted of listeners who were less than 36 years old. Both groups had listeners with normal hearing as well as listeners with mild to moderate sensorineural loss. The masking paradigm consisted of a continuous low-pass-filtered (1000-Hz) noise, which was mixed with the output of a self-tracking, sweep-frequency Bekesy audiometer. Thresholds were measured in quiet and with maskers at 70 and 90 dB SPL. The upward-masked thresholds were similar for young and elderly hearing-impaired listeners. A few elderly listeners had lower upward-masked thresholds compared with the young control group; however, their on-frequency masked thresholds were nearly identical to the control group. A significant correlation was found between upward-masked thresholds and the Speech Perception in Noise (SPIN) test in elderly listeners.     

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.     

Recognition of filtered words in noise at higher-than-normal levels: decreases in scores with and without increases in masking

To examine spectral effects on declines in speech recognition in noise at high levels, word recognition for 18 young adults with normal hearing was assessed for low-pass-filtered speech and speech-shaped maskers or high-pass-filtered speech and speech-shaped maskers at three speech levels (70, 77, and 84 dB SPL) for each of three signal-to-noise ratios (+8, +3, and -2 dB). An additional low-level noise produced equivalent masked thresholds for all subjects. Pure-tone thresholds were measured in quiet and in all maskers. If word recognition was determined entirely by signal-to-noise ratio, and was independent of signal levels and the spectral content of speech and maskers, scores should remain constant with increasing level for both low- and high-frequency speech and maskers. Recognition of low-frequency speech in low-frequency maskers and high-frequency speech in high-frequency maskers decreased significantly with increasing speech level when signal-to-noise ratio was held constant. For low-frequency speech and speech-shaped maskers, the decline was attributed to nonlinear growth of masking which reduced the "effective" signal-to-noise ratio at high levels, similar to previous results for broadband speech and speech-shaped maskers. Masking growth and reduced "effective" signal-to-noise ratio accounted for some but not all the decline in recognition of high-frequency speech in high-frequency maskers.     

On the relation between the growth of loudness and the discrimination of intensity for pure tones

The intensity jnd is often assumed to depend on the slope of the loudness function. One way to test this assumption is to measure the jnd for a sound that falls on distinctly different loudness functions. Two such functions were generated by presenting a 1000-Hz tone in narrow-band noise (925-1080 Hz) set at 70 dB SPL and in wideband noise (75-9600 Hz) set at 80 dB SPL. Over a range from near threshold to about 75 dB SPL, the loudness function for the tone is much steeper in the narrow-band noise than in the wideband noise. At 72 dB SPL, where the two loudness curves cross, the tone's jnd was measured in each noise by a block up-down two-interval forced-choice procedure. Despite the differences in slope (and in sensation level), the jnd (delta I/I) is nearly the same in the two noises, 0.22 in narrow-band noise and 0.20 in wideband noise. The mean value of 0.21 is close to the value of 0.25 interpolated from Jesteadt et al. [J. Acoust. Soc. Am. 61, 169-176 (1977)] for a 1000-Hz tone that had the same loudness in quiet as did our 72-dB tone in noise, but lay on a loudness function with a much lower slope. These and other data demonstrate that intensity discrimination for pure tones is unrelated to the slope of the loudness function.     

Excess masking among listeners with a sensorineural hearing loss

Three experiments were conducted to determine whether listeners with a sensorineural hearing loss exhibited greater than normal amounts of masking at frequencies above the frequency of the masker. Excess masking was defined as the difference (in dB) between the masked thresholds actually obtained from a hearing-impaired listener and the expected thresholds calculated for the same individual. The expected thresholds were the power sum of the listener's thresholds in quiet and the average masked thresholds obtained from a group of normal-hearing subjects at the test frequency. Hearing-impaired listeners, with thresholds in quiet ranging from approximately 35-70 dB SPL (at test frequencies between 500-3000 Hz), displayed approximately 12-15 dB of maximum excess masking. The maximum amount of excess masking occurred in the region where the threshold in quiet of the hearing-impaired listener and the average normal masked threshold were equal. These findings indicate that listeners with a sensorineural hearing loss display one form of reduced frequency selectivity (i.e., abnormal upward spread of masking) even when their thresholds in quiet are taken into account.     

Combined spectral and temporal enhancement to improve cochlear-implant speech perception

The present study examined the effect of combined spectral and temporal enhancement on speech recognition by cochlear-implant (CI) users in quiet and in noise. The spectral enhancement was achieved by expanding the short-term Fourier amplitudes in the input signal. Additionally, a variation of the Transient Emphasis Spectral Maxima (TESM) strategy was applied to enhance the short-duration consonant cues that are otherwise suppressed when processed with spectral expansion. Nine CI users were tested on phoneme recognition tasks and ten CI users were tested on sentence recognition tasks both in quiet and in steady, speech-spectrum-shaped noise. Vowel and consonant recognition in noise were significantly improved with spectral expansion combined with TESM. Sentence recognition improved with both spectral expansion and spectral expansion combined with TESM. The amount of improvement varied with individual CI users. Overall the present results suggest that customized processing is needed to optimize performance according to not only individual users but also listening conditions.     

The effect of masker level uncertainty on intensity discrimination

Thresholds were measured for detection of an increment in level of a 60-dB SPL target tone at 1 kHz, either in quiet or in the presence of maskers at 0.5 and 2 kHz. Interval-by-interval level rove applied independently to remote masker tones substantially elevated thresholds compared to intensity discrimination in quiet, an effect on the order of 10+dB [10 log(DeltaII)]. Asynchronous onset and stimulus envelope mismatches across frequency reduced but did not eliminate masking. A preinterval cue to signal frequency had no effect, but cuing masker frequency reduced thresholds, whether or not masker level was also cued. About 1 to 2 dB of threshold elevation in these conditions can be attributed to energetic masking. Decreasing the overall presentation level and increasing masker separation essentially eliminates energetic masking; under these conditions masker level rove elevates thresholds by approximately 7 dB when the target and masker tones are gated synchronously. This masking persists even when the flanking masker tones are presented contralateral to the target. Results suggest that observers tend to listen synthetically, even in conditions when this strategy reduces sensitivity to the intensity increment.     

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.     

Spectral and threshold effects on recognition of speech at higher-than-normal levels

To examine spectral and threshold effects for speech and noise at high levels, recognition of nonsense syllables was assessed for low-pass-filtered speech and speech-shaped maskers and high-pass-filtered speech and speech-shaped maskers at three speech levels, with signal-to-noise ratio held constant. Subjects were younger adults with normal hearing and older adults with normal hearing but significantly higher average quiet thresholds. A broadband masker was always present to minimize audibility differences between subject groups and across presentation levels. For subjects with lower thresholds, the declines in recognition of low-frequency syllables in low-frequency maskers were attributed to nonlinear growth of masking which reduced "effective" signal-to-noise ratio at high levels, whereas the decline for subjects with higher thresholds was not fully explained by nonlinear masking growth. For all subjects, masking growth did not entirely account for declines in recognition of high-frequency syllables in high-frequency maskers at high levels. Relative to younger subjects with normal hearing and lower quiet thresholds, older subjects with normal hearing and higher quiet thresholds had poorer consonant recognition in noise, especially for high-frequency speech in high-frequency maskers. Age-related effects on thresholds and task proficiency may be determining factors in the recognition of speech in noise at high levels.     

一种抑制方向性噪声的双耳语音增强算法

为了给双耳听力设备佩戴者带来更好的语音可懂度,提出了一种利用双耳时间差与声级差的近场语音增强算法,该方法首先利用这两种差异来估计语音的功率谱和语音的相干函数,然后计算干扰噪声在左右耳间的头相关传输函数的比值,最后构造两个维纳滤波器。客观评价的参数显示该算法去噪效果优于对比算法而目标语音的时间差误差和声级差误差低于对比算法。主观的言语接受阈测试表明该方法能有效提高语音可懂度。结果表明,该算法在能够有效去除干扰噪声的同时,保留了目标语音的空间信息。      

A comparison of underwater hearing sensitivity in bottlenose dolphins (Tursiops truncatus) determined by electrophysiological and behavioral methods

Variable stimulus presentation methods are used in auditory evoked potential (AEP) estimates of cetacean hearing sensitivity, each of which might affect stimulus reception and hearing threshold estimates. This study quantifies differences in underwater hearing thresholds obtained by AEP and behavioral means. For AEP estimates, a transducer embedded in a suction cup (jawphone) was coupled to the dolphin's lower jaw for stimulus presentation. Underwater AEP thresholds were obtained for three dolphins in San Diego Bay and for one dolphin in a quiet pool. Thresholds were estimated from the envelope following response at carrier frequencies ranging from 10 to 150 kHz. One animal, with an atypical audiogram, demonstrated significantly greater hearing loss in the right ear than in the left. Across test conditions, the range and average difference between AEP and behavioral threshold estimates were consistent with published comparisons between underwater behavioral and in-air AEP thresholds. AEP thresholds for one animal obtained in-air and in a quiet pool demonstrated a range of differences of -10 to 9 dB (mean = 3 dB). Results suggest that for the frequencies tested, the presentation of sound stimuli through a jawphone, underwater and in-air, results in acceptable differences to AEP threshold estimates.