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.     

Factors influencing intelligibility of ideal binary-masked speech: implications for noise reduction

The application of the ideal binary mask to an auditory mixture has been shown to yield substantial improvements in intelligibility. This mask is commonly applied to the time-frequency (T-F) representation of a mixture signal and eliminates portions of a signal below a signal-to-noise-ratio (SNR) threshold while allowing others to pass through intact. The factors influencing intelligibility of ideal binary-masked speech are not well understood and are examined in the present study. Specifically, the effects of the local SNR threshold, input SNR level, masker type, and errors introduced in estimating the ideal mask are examined. Consistent with previous studies, intelligibility of binary-masked stimuli is quite high even at -10 dB SNR for all maskers tested. Performance was affected the most when the masker dominated T-F units were wrongly labeled as target-dominated T-F units. Performance plateaued near 100% correct for SNR thresholds ranging from -20 to 5 dB. The existence of the plateau region suggests that it is the pattern of the ideal binary mask that matters the most rather than the local SNR of each T-F unit. This pattern directs the listener's attention to where the target is and enables them to segregate speech effectively in multitalker environments.     

Subspace algorithms for noise reduction in cochlear implants

A single-channel algorithm is proposed for noise reduction in cochlear implants. The proposed algorithm is based on subspace principles and projects the noisy speech vector onto "signal" and "noise" subspaces. An estimate of the clean signal is made by retaining only the components in the signal subspace. The performance of the subspace reduction algorithm is evaluated using 14 subjects wearing the Clarion device. Results indicated that the subspace algorithm produced significant improvements in sentence recognition scores compared to the subjects' daily strategy, at least in stationary noise. Further work is needed to extend the subspace algorithm to nonstationary noise environments.     

高层住宅阳台降噪量的计算分析

高层住宅中的阳台不仅具有居住及建筑上的功能,对道路交通噪声也有衰减作用。本文通过建立噪声传播模型计算高层住宅阳台的插入损失,并探讨了影响阳台降噪量的一些因素。同时对阳台内部吸声处理的效果进行了理论预测。     

A study of noise reduction by acoustic shock waves

I.IntroductionAcousticshockwavcs(ASW)isanimportantphcnomcnoninnonlinearacoustics.Experimentalrcsultshavcshownthatwhenanaircraftcngincinletopcratesneartheson-iccondition,vcrystrongnoisegcncratedbythcfanscanbcreduccdgreat1yowingtothcformationofASWatthcthroatofthcin1etll].ASWisadiscontinuityofacousticvaria-bles,whichisdifTcrcntfromthcshockwavesoccurringinhighspcedsteadyflowinducts.Theformer'sintensityismuch1cssthanthelattcr's.Furthcrmorc,thepositionandintensityofASWisalwayschangedwithtime.l…     

声激波降噪研究

本文数值研究了声激波产生的条件及其降噪效果,详细讨论了声源强度,频率,管道形状,喉部流动马赫数及多频声源对声激波降噪效果的影响。由于声激波降噪需要附加一段具有高速流动的缩放管道,它会附加流动噪声和压力损失,本文提供了这方面的一组实验数据,以说明应用声激波消声器的现实可能性。     

A comparative study of phase-shifting algorithms in digital speckle pattern interferometry

Digital speckle pattern interferometry (DSPI) is a tool for making qualitative as well as quantitative measurements of deformation of objects. Phase-shifting algorithms in DSPI are useful for extracting quantitative deformation data from the system. Comparative studies of the different phase-shifting algorithms in DSPI for object deformation measurement are presented. Static and quasi-dynamic deformation of the object can be measured using these algorithms. Error compensating five-step phase-shifting method is used for the algorithms.     

Speech intelligibility improvements with hearing aids using bilateral and binaural adaptive multichannel Wiener filtering based noise reduction

This paper evaluates noise reduction techniques in bilateral and binaural hearing aids. Adaptive implementations (on a real-time test platform) of the bilateral and binaural speech distortion weighted multichannel Wiener filter (SDW-MWF) and a competing bilateral fixed beamformer are evaluated. As the SDW-MWF relies on a voice activity detector (VAD), a realistic binaural VAD is also included. The test subjects (both normal hearing subjects and hearing aid users) are tested by an adaptive speech reception threshold (SRT) test in different spatial scenarios, including a realistic cafeteria scenario with nonstationary noise. The main conclusions are: (a) The binaural SDW-MWF can further improve the SRT (up to 2 dB) over the improvements achieved by bilateral algorithms, although a significant difference is only achievable if the binaural SDW-MWF uses a perfect VAD. However, in the cafeteria scenario only the binaural SDW-MWF achieves a significant SRT improvement (2.6 dB with perfect VAD, 2.2 dB with real VAD), for the group of hearing aid users. (b) There is no significant degradation when using a real VAD at the input signal-to-noise ratio (SNR) levels where the hearing aid users reach their SRT. (c) The bilateral SDW-MWF achieves no SRT improvements compared to the bilateral fixed beamformer.     

Experimental evaluation of leaky least-mean-square algorithms for active noise reduction in communication headsets

An adaptive leaky normalized least-mean-square (NLMS) algorithm has been developed to optimize stability and performance of active noise cancellation systems. The research addresses LMS filter performance issues related to insufficient excitation, nonstationary noise fields, and time-varying signal-to-noise ratio. The adaptive leaky NLMS algorithm is based on a Lyapunov tuning approach in which three candidate algorithms, each of which is a function of the instantaneous measured reference input, measurement noise variance, and filter length, are shown to provide varying degrees of tradeoff between stability and noise reduction performance. Each algorithm is evaluated experimentally for reduction of low frequency noise in communication headsets, and stability and noise reduction performance are compared with that of traditional NLMS and fixed-leakage NLMS algorithms. Acoustic measurements are made in a specially designed acoustic test cell which is based on the original work of Ryan et al. ["Enclosure for low frequency assessment of active noise reducing circumaural headsets and hearing protection," Can. Acoust. 21, 19-20 (1993)] and which provides a highly controlled and uniform acoustic environment. The stability and performance of the active noise reduction system, including a prototype communication headset, are investigated for a variety of noise sources ranging from stationary tonal noise to highly nonstationary measured F-16 aircraft noise over a 20 dB dynamic range. Results demonstrate significant improvements in stability of Lyapunov-tuned LMS algorithms over traditional leaky or nonleaky normalized algorithms, while providing noise reduction performance equivalent to that of the NLMS algorithm for idealized noise fields.     

Binaural speech intelligibility in noise for hearing-impaired listeners

The effect of head-induced interaural time delay (ITD) and interaural level differences (ILD) on binaural speech intelligibility in noise was studied for listeners with symmetrical and asymmetrical sensorineural hearing losses. The material, recorded with a KEMAR manikin in an anechoic room, consisted of speech, presented from the front (0 degree), and noise, presented at azimuths of 0 degree, 30 degrees, and 90 degrees. Derived noise signals, containing either only ITD or only ILD, were generated using a computer. For both groups of subjects, speech-reception thresholds (SRT) for sentences in noise were determined as a function of: (1) noise azimuth, (2) binaural cue, and (3) an interaural difference in overall presentation level, simulating the effect of a monaural hearing acid. Comparison of the mean results with corresponding data obtained previously from normal-hearing listeners shows that the hearing impaired have a 2.5 dB higher SRT in noise when both speech and noise are presented from the front, and 2.6-5.1 dB less binaural gain when the noise azimuth is changed from 0 degree to 90 degrees. The gain due to ILD varies among the hearing-impaired listeners between 0 dB and normal values of 7 dB or more. It depends on the high-frequency hearing loss at the side presented with the most favorable signal-to-noise (S/N) ratio. The gain due to ITD is nearly normal for the symmetrically impaired (4.2 dB, compared with 4.7 dB for the normal hearing), but only 2.5 dB in the case of asymmetrical impairment. When ITD is introduced in noise already containing ILD, the resulting gain is 2-2.5 dB for all groups. The only marked effect of the interaural difference in overall presentation level is a reduction of the gain due to ILD when the level at the ear with the better S/N ratio is decreased. This implies that an optimal monaural hearing aid (with a moderate gain) will hardly interfere with unmasking through ITD, while it may increase the gain due to ILD by preventing or diminishing threshold effects.     

一种增强带噪语音可懂度的新算法

一般的语音增强算法在强噪声环境中只能提高信噪比,不能提高可懂度。本文提出用可调节白噪声代替信号中非语音部分的语音可懂度增强处理新算法。实验证明此方法能明显改善强噪声时的语音可懂度,能对低至-10dB的带噪语音信号进行有效的可懂度增强。     

噪声对单耳语言可懂度的影响

单耳通信时,周边噪声对语言可懂度产生影响。针对信号侧语音信号强度70dB时,研究3种不同类型噪声下,干扰侧不同强度噪声和信号侧不同信噪比情况的语言可懂度。实验结果表明:当信号侧信噪比大于某一阈值时,干扰侧噪声对可懂度不产生显著影响,该阈值同噪声类型有关;而在信号侧低信噪比的情形下,干扰侧适当强度噪声可提高信号侧语言可懂度,最佳干扰噪声级为78—82dB,过大的干扰侧噪声级导致可懂度下降。基于心理声学和生理学的初步机理发现:噪声环境下的语音识别中,对侧耳中耳肌肉伸缩对噪声感知的抑制提高了信号侧语言可懂度。     

流动管道内反声降噪数值研究

本文用流体力学方法建立了流动管道内反声降噪数值方法,其中,严格推导了反声源的数学模型.本文还给出了产逆流和声顺流传播时管内声场反声影响的数值算例,并讨论了流动速度、反声源强度和相位的影响.算例表明:反声降噪具有很好的工程应用前景.     

A numerical study of noise reduction in ducts with flow by anti-sound

I.IntroductionAnti-soundisalsocal1edactivenoisecontro1(ANC).Itsbasicidea,presentedintheLueg'spatentinl936l'l,isthatthenoisereductionisobtainedbyuseofthesignalpro-cessingofthepreliminarysoundsource(i.e.noisesource)toformcoherenceinfluencebe-twccnthepreliminarysoundsourceandthesecondarysoundsourcc(i.c.anti-soundsource).Therearesomeadvantagesofanti-soundsuchasactivecontro1,lcsseffectonthecharacteristicsofnoisesourccandmorereductionoflowfrequcncynoisc.Inrecentyears,therewerealotoftheoreticalande…     

Neural correlates of intelligibility in speech investigated with noise vocoded speech--a positron emission tomography study

Functional imaging studies of speech perception in the human brain have identified a key role for auditory association areas in the temporal lobes (bilateral superior temporal gyri and sulci) in the perceptual processing of the speech signal. This is extended to suggest some functional specialization within this bilateral system, with a particular role for the left anterior superior temporal sulcus (STS) in processing intelligible speech. In the current study, noise-vocoded speech was used to vary the intelligibility of speech parametrically. This replicated the finding of a selective response to intelligibility in speech in the left anterior superior temporal sulcus, in contrast to the posterior superior temporal sulcus, which showed a response profile insensitive to the degree of intelligibility. These results are related to theories of functional organization in the human auditory system, which have indicated that there are separate processing streams, with different functional roles, running anterior and posterior to primary auditory cortex. Specifically, it is suggested that an anterior stream processing intelligibility can be distinguished from a posterior stream associated with transient representations, important in spoken repetition and working memory.     

单通道语音增强算法对汉语语音可懂度影响的研究

考察了当前常用的几种单通道语音增强算法对汉语语音可懂度的影响。受不同类型噪音干扰的语音经过5种单通道语音增强算法的处理后,播放给具有正常听力水平的被试进行听辩,考察增强后语音的可懂度。实验结果表明,语音增强算法并不能改进语音的可懂度水平;通过分析具体的错误原因,发现听辩错误主要来自于音素错误,与声调关系不大;而且,同英文的辨识结果相比,一些增强算法对于中、英文可懂度影响差异显著。