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.     

The effects of multichannel compression/expansion amplification on the intelligibility of nonsense syllables in noise

The effects of six-channel compression and expansion amplification on the intelligibility of nonsense syllables embedded in speech spectrum noise were examined for four hearing-impaired subjects. For one condition (linear) the stimulus was given six-channel amplification with frequency shaping to suit the subject's hearing loss. The other condition (nonlinear) was the same except that low level inputs, to any given channel, received expansion amplification and high level inputs received compression. For each condition, each subject received the nonsense syllables at three different input levels, representing low, average, and high intensity speech. The results of this study, like those of most other studies of multichannel compression, are mainly negative. Nonlinear processing (mainly expansion) of low intensity speech resulted in a significant degradation of speech intelligibility for two subjects and in no improvement for the others. One subject showed a significant improvement in intelligibility for the nonlinearly processed average intensity speech and another subject showed significant improvement for the high intensity input (mainly compression). Clearly, nonlinear processing is beneficial for some subjects, under some listening conditions, but further research is needed to identify the relevent characteristics of such subjects. An acoustic analysis of selected items revealed that the failure of expansion to improve intelligibility was primarily due to the very low intensity consonants /e/ and /k/, in final position, being presented at an even lower intensity in the expansion condition than in the linear condition. Expansion may be worth further investigation with different parameters. Several other problems caused by the multichannel processing were also revealed. These included alteration of spectral shapes and band interaction effects. Ways of overcoming these problems, and of capitalizing on the likely advantages of multichannel amplification, are currently being investigated.     

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.     

Principal-component amplitude compression for the hearing impaired

Principal-component amplitude compression, a means for matching speech to the reduced dynamic range in sensorineural hearing impairments, is a multiband approach aimed at preserving details of spectral shape while reducing overall level variation. The effect of compression has been studied for the first and second principal components (PC1 an PC2) of the short-term speech spectrum, which are roughly representative of overall level and spectral tilt, respectively. Compression of PC1 roughly equalizes consonant and vowel levels while compression of PC2 provides time-varying high-frequency emphasis. The effect on speech intelligibility of sensorineural hearing-impaired listeners of two principal-component compression system implementations, compression of PC1 and compression of both PC1 and PC2, was compared to that of linear amplification (LA), independent compression of multiple bands (MBC), and wideband compression (WC). Results indicate that compression of overall level as provided by compression of PC1 and WC improved intelligibility relative to LA over a 10- to 15-dB range of input levels. While MBC was beneficial in some cases, it did not provide higher intelligibility than WC. Compression of PC2 did not benefit but rather degraded performance relative to LA. Error analyses and band-level measurements indicate that the highest intelligibility is obtained when audibility is improved and the relative spectral shapes of different speech sounds are preserved.     

An analysis of the effects of electrical field interaction with an acoustic model of cochlear implants

Electrical field interaction caused by current spread in a cochlear implant was modeled in an explicit way in an acoustic model (the SPREAD model) presented to six listeners with normal hearing. The typical processing of cochlear implants was modeled more closely than in traditional acoustic models by careful selection of parameters related to current spread or parameters that could amplify the electrical field interactions caused by current spread. These parameters were the insertion depth, electrode spacing, electrical dynamic range, and dynamic range compression function. The hypothesis was that current spread could account for the asymptote in performance in speech intelligibility experiments observed at around seven stimulation channels in a number of cochlear implant studies. Speech intelligibility for sentences, vowels, and consonants at three noise levels (SNR of +15 dB, +10 dB, and +5 dB) was measured as a function of the number of spectral channels (4, 7, and 16). The SPREAD model appears to explain the asymptote in speech intelligibility at seven channels for all noise levels for all speech material used in this study. It is shown that the compressive amplitude mapping used in cochlear implants can have a detrimental effect on the number of effective channels.     

On application of adaptive decorrelation filtering to assistive listening

This paper describes an application of the multichannel signal processing technique of adaptive decorrelation filtering to the design of an assistive listening system. A simulated "dinner table" scenario was studied. The speech signal of a desired talker was corrupted by three simultaneous speech jammers and by a speech-shaped diffusive noise. The technique of adaptive decorrelation filtering processing was used to extract the desired speech from the interference speech and noise. The effectiveness of the assistive listening system was evaluated by observing improvements in A-weighted signal-to-noise ratio (SNR) and in sentence intelligibility, where the latter was evaluated in a listening test with eight normal hearing subjects and three subjects with hearing impairments. Significant improvements in SNR and sentence intelligibility were achieved with the use of the assistive listening system. For subjects with normal hearing, the speech reception threshold was improved by 3 to 5 dBA, and for subjects with hearing impairments, the threshold was improved by 4 to 8 dBA.     

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.     

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

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

Effects of directional microphone and adaptive multichannel noise reduction algorithm on cochlear implant performance

Although cochlear implant (CI) users have enjoyed good speech recognition in quiet, they still have difficulties understanding speech in noise. We conducted three experiments to determine whether a directional microphone and an adaptive multichannel noise reduction algorithm could enhance CI performance in noise and whether Speech Transmission Index (STI) can be used to predict CI performance in various acoustic and signal processing conditions. In Experiment I, CI users listened to speech in noise processed by 4 hearing aid settings: omni-directional microphone, omni-directional microphone plus noise reduction, directional microphone, and directional microphone plus noise reduction. The directional microphone significantly improved speech recognition in noise. Both directional microphone and noise reduction algorithm improved overall preference. In Experiment II, normal hearing individuals listened to the recorded speech produced by 4- or 8-channel CI simulations. The 8-channel simulation yielded similar speech recognition results as in Experiment I, whereas the 4-channel simulation produced no significant difference among the 4 settings. In Experiment III, we examined the relationship between STIs and speech recognition. The results suggested that STI could predict actual and simulated CI speech intelligibility with acoustic degradation and the directional microphone, but not the noise reduction algorithm. Implications for intelligibility enhancement are discussed.     

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.     

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.     

数字助听器中广义旁瓣抵消器结构的汉语语音处理技术

自适应波束形成算法是数字助听器的核心算法之一。针对自适应波束形成算法中不可避免存在的语音泄漏,本文先对传统GSC结构自适应波束形成算法进行理论研究,并提出一种汉语处理技术,补偿泄漏的语音。这种汉语处理技术利用汉语语音特有的基音频率信息,调整语音幅度谱包络,提高谱包络与基频曲线形状的相似度以提高语音的可懂度。针对泄漏的语音在高频清辅音段有较大损失的特点,在频域上对清辅音进行放大,在不改变共振峰结构的情况下,提高清辅音的能量,同时降低语音间隔段GSC算法泄漏的噪声能量,提高对语音的辨别。仿真实验结果表明,这种汉语语音处理能够补偿自适应波束形成算法造成的语音泄漏,提高语音的可懂度。      

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.     

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.     

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.     

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.     

Effect of the speed of a single-channel dynamic range compressor on intelligibility in a competing speech task

Using a "noise-vocoder" cochlear implant simulator [Shannon et al., Science 270, 303-304 (1995)], the effect of the speed of dynamic range compression on speech intelligibility was assessed, using normal-hearing subjects. The target speech had a level 5 dB above that of the competing speech. Initially, baseline performance was measured with no compression active, using between 4 and 16 processing channels. Then, performance was measured using a fast-acting compressor and a slow-acting compressor, each operating prior to the vocoder simulation. The fast system produced significant gain variation over syllabic timescales. The slow system produced significant gain variation only over the timescale of sentences. With no compression active, about six channels were necessary to achieve 50% correct identification of words in sentences. Sixteen channels produced near-maximum performance. Slow-acting compression produced no significant degradation relative to the baseline. However, fast-acting compression consistently reduced performance relative to that for the baseline, over a wide range of performance levels. It is suggested that fast-acting compression degrades performance for two reasons: (1) because it introduces correlated fluctuations in amplitude in different frequency bands, which tends to produce perceptual fusion of the target and background sounds and (2) because it reduces amplitude modulation depth and intensity contrasts.     

A detailed study on the effects of noise on speech intelligibility

A wavelet representation of speech was used to display the instantaneous amplitude and phase within 14 octave frequency bands, representing the envelope and the carrier within each band. Adding stationary noise alters the wavelet pattern, which can be understood as a combination of three simultaneously occurring subeffects: two effects on the wavelet levels (one systematic and one stochastic) and one effect on the wavelet phases. Specific types of signal processing were applied to speech, which allowed each effect to be either included or excluded. The impact of each effect (and of combinations) on speech intelligibility was measured with CVC's. It appeared that the systematic level effect (i.e., the increase of each speech wavelet intensity with the mean noise intensity) has the most degrading effect on speech intelligibility, which is in accordance with measures such as the modulation transfer function and the speech transmission index. However, also the introduction of stochastic level fluctuations and disturbance of the carrier phase seriously contribute to reduced intelligibility in noise. It is argued that these stochastic effects are responsible for the limited success of spectral subtraction as a means to improve speech intelligibility. Results can provide clues for effective noise suppression with respect to intelligibility.     

Performance of hearing-impaired listeners under various types of amplitude compression

Speech perception by subjects with sensorineural hearing impairment was studied using various types of short-term (syllabic) amplitude compression. Average speech level was approximately constant. In quiet, a single-channel wideband compression (WBC) with compression ratio equal to 10, attack time 10 ms and release time 90 ms produced significantly higher scores than a three-channel multiband compression (MBC) or no compression when a nonsense syllable test (City University of New York) was used. The scores under MBC, WBC, or no compression were not significantly different when the modified rhyme test (MRT) was used. But when overshoots caused by compression were clipped, the MRT scores improved significantly. The influence of MBC on reverberant speech and of WBC on noisy speech were tested with the MRT. Reverberation reduced the scores, and this reduction was the same with compression as without. Noise added to speech before compression also reduced the scores, but the reduction was larger with compression than without. When noise was added after compression, an improvement was observed when WBC had a compression ratio of about 5, attack time 1 ms, and release time 30 ms. Other compression modes (e.g., with high-frequency pre-emphasis) did not show an improvement. The results indicate that WBC with a compression ratio around 5, attack time shorter than 3 ms, and release time between 30 and 90 ms can be beneficial if signal-to-noise ratio is large, or, if in a noisy or reverberant environment, the effects of noise or reverberation are eliminated by using listening systems.     

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.