The effect of overlap-masking on binaural reverberant word intelligibility

Reverberation interferes with the ability to understand speech in rooms. Overlap-masking explains this degradation by assuming reverberant phonemes endure in time and mask subsequent reverberant phonemes. Most listeners benefit from binaural listening when reverberation exists, indicating that the listener's binaural system processes the two channels to reduce the reverberation. This paper investigates the hypothesis that the binaural word intelligibility advantage found in reverberation is a result of binaural overlap-masking release with the reverberation acting as masking noise. The tests utilize phonetically balanced word lists (ANSI-S3.2 1989), that are presented diotically and binaurally with recorded reverberation and reverberation-like noise. A small room, 62 m3, reverberates the words. These are recorded using two microphones without additional noise sources. The reverberation-like noise is a modified form of these recordings and has a similar spectral content. It does not contain binaural localization cues due to a phase randomization procedure. Listening to the reverberant words binaurally improves the intelligibility by 6.0% over diotic listening. The binaural intelligibility advantage for reverberation-like noise is only 2.6%. This indicates that binaural overlap-masking release is insufficient to explain the entire binaural word intelligibility advantage in reverberation.     

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.     

Binaural prediction of speech intelligibility in reverberant rooms with multiple noise sources

When speech is in competition with interfering sources in rooms, monaural indicators of intelligibility fail to take account of the listener's abilities to separate target speech from interfering sounds using the binaural system. In order to incorporate these segregation abilities and their susceptibility to reverberation, Lavandier and Culling [J. Acoust. Soc. Am. 127, 387-399 (2010)] proposed a model which combines effects of better-ear listening and binaural unmasking. A computationally efficient version of this model is evaluated here under more realistic conditions that include head shadow, multiple stationary noise sources, and real-room acoustics. Three experiments are presented in which speech reception thresholds were measured in the presence of one to three interferers using real-room listening over headphones, simulated by convolving anechoic stimuli with binaural room impulse-responses measured with dummy-head transducers in five rooms. Without fitting any parameter of the model, there was close correspondence between measured and predicted differences in threshold across all tested conditions. The model's components of better-ear listening and binaural unmasking were validated both in isolation and in combination. The computational efficiency of this prediction method allows the generation of complex "intelligibility maps" from room designs.     

Multichannel speech intelligibility and talker recognition using monaural, binaural, and three-dimensional auditory presentation

In a 3D auditory display, sounds are presented over headphones in a way that they seem to originate from virtual sources in a space around the listener. This paper describes a study on the possible merits of such a display for bandlimited speech with respect to intelligibility and talker recognition against a background of competing voices. Different conditions were investigated: speech material (words/sentences), presentation mode (monaural/binaural/3D), number of competing talkers (1-4), and virtual position of the talkers (in 45 degrees-steps around the front horizontal plane). Average results for 12 listeners show an increase of speech intelligibility for 3D presentation for two or more competing talkers compared to conventional binaural presentation. The ability to recognize a talker is slightly better and the time required for recognition is significantly shorter for 3D presentation in the presence of two or three competing talkers. Although absolute localization of a talker is rather poor, spatial separation appears to have a significant effect on communication. For either speech intelligibility, talker recognition, or localization, no difference is found between the use of an individualized 3D auditory display and a general display.     

Improvement of intelligibility of ideal binary-masked noisy speech by adding background noise

When a target-speech/masker mixture is processed with the signal-separation technique, ideal binary mask (IBM), intelligibility of target speech is remarkably improved in both normal-hearing listeners and hearing-impaired listeners. Intelligibility of speech can also be improved by filling in speech gaps with un-modulated broadband noise. This study investigated whether intelligibility of target speech in the IBM-treated target-speech/masker mixture can be further improved by adding a broadband-noise background. The results of this study show that following the IBM manipulation, which remarkably released target speech from speech-spectrum noise, foreign-speech, or native-speech masking (experiment 1), adding a broadband-noise background with the signal-to-noise ratio no less than 4 dB significantly improved intelligibility of target speech when the masker was either noise (experiment 2) or speech (experiment 3). The results suggest that since adding the noise background shallows the areas of silence in the time-frequency domain of the IBM-treated target-speech/masker mixture, the abruption of transient changes in the mixture is smoothed and the perceived continuity of target-speech components becomes enhanced, leading to improved target-speech intelligibility. The findings are useful for advancing computational auditory scene analysis, hearing-aid/cochlear-implant designs, and understanding of speech perception under "cocktail-party" conditions.     

提高耳语音可懂度的非对称压缩语音增强方法

提出两种基于非对称代价函数的耳语音增强算法,将语音增强过程中的放大失真和压缩失真区分对待。Modified ItakuraSaito (MIS)算法对放大失真给予更多的惩罚,而Kullback-Leibler (KL)算法则对压缩失真给予更多的惩罚。实验结果表明,在低于—6 dB的低信噪比情况中,经MIS算法增强后的耳语音的可懂度相比传统算法有显著提高;而KL算法则获得了同最小均方误差语音增强算法近似的可懂度提高效果,证实了耳语音中的放大失真和压缩失真对于耳语音可懂度的影响并不相同,低信噪比时较大的压缩失真有助于提高耳语音可懂度,而高信噪比时的压缩失真对耳语音可懂度影响较小。     

Prediction of the influence of reverberation on binaural speech intelligibility in noise and in quiet

Reverberation usually degrades speech intelligibility for spatially separated speech and noise sources since spatial unmasking is reduced and late reflections decrease the fidelity of the received speech signal. The latter effect could not satisfactorily be predicted by a recently presented binaural speech intelligibility model [Beutelmann et al. (2010). J. Acoust. Soc. Am. 127, 2479-2497]. This study therefore evaluated three extensions of the model to improve its predictions: (1) an extension of the speech intelligibility index based on modulation transfer functions, (2) a correction factor based on the room acoustical quantity "definition," and (3) a separation of the speech signal into useful and detrimental parts. The predictions were compared to results of two experiments in which speech reception thresholds were measured in a reverberant room in quiet and in the presence of a noise source for listeners with normal hearing. All extensions yielded better predictions than the original model when the influence of reverberation was strong, while predictions were similar for conditions with less reverberation. Although model (3) differed substantially in the assumed interaction of binaural processing and early reflections, its predictions were very similar to model (2) that achieved the best fit to the data.     

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.     

An evaluation of objective measures for intelligibility prediction of time-frequency weighted noisy speech

Existing objective speech-intelligibility measures are suitable for several types of degradation, however, it turns out that they are less appropriate in cases where noisy speech is processed by a time-frequency weighting. To this end, an extensive evaluation is presented of objective measure for intelligibility prediction of noisy speech processed with a technique called ideal time frequency (TF) segregation. In total 17 measures are evaluated, including four advanced speech-intelligibility measures (CSII, CSTI, NSEC, DAU), the advanced speech-quality measure (PESQ), and several frame-based measures (e.g., SSNR). Furthermore, several additional measures are proposed. The study comprised a total number of 168 different TF-weightings, including unprocessed noisy speech. Out of all measures, the proposed frame-based measure MCC gave the best results (ρ?=?0.93). An additional experiment shows that the good performing measures in this study also show high correlation with the intelligibility of single-channel noise reduced speech.     

The impact of reverberant self-masking and overlap-masking effects on speech intelligibility by cochlear implant listeners (L)

The purpose of this study is to determine the relative impact of reverberant self-masking and overlap-masking effects on speech intelligibility by cochlear implant listeners. Sentences were presented in two conditions wherein reverberant consonant segments were replaced with clean consonants, and in another condition wherein reverberant vowel segments were replaced with clean vowels. The underlying assumption is that self-masking effects would dominate in the first condition, whereas overlap-masking effects would dominate in the second condition. Results indicated that the degradation of speech intelligibility in reverberant conditions is caused primarily by self-masking effects that give rise to flattened formant transitions.     

Predictors of speech intelligibility in rooms

Three different types of acoustical measures were compared as predictors of speech intelligibility in rooms of varied size and acoustical conditions. These included signal-to-noise measures, the speech transmission index derived from modulation transfer functions, and useful/detrimental sound ratios obtained from early/late sound ratios, speech, and background levels. The most successful forms of each type of measure were of similar prediction accuracy, but the useful/detrimental ratios based on a 0.08-s early time interval were most accurate. Several physical measures, although based on very different calculation procedures, were quite strongly related to each other.     

自适应零限波束形成语音增强算法鲁棒性分析

分析了双传声器自适应零限波束形成语音增强算法对传声器不一致和本底噪声的鲁棒性。结果表明:信干比越高,算法对传声器不一致的鲁棒性越差;当信干比很低时,算法对相位不一致是鲁棒的。幅度不一致会降低算法对干扰的抑制能力,但引起的目标信号失真很小。相对于传声器不一致,本底噪声对算法性能的影响较小。对自适应滤波器权值加约束和对传声器做校准可以提高算法性能。     

A blind algorithm for reverberation-time estimation using subband decomposition of speech signals

An algorithm for blind estimation of reverberation time (RT) in speech signals is proposed. Analysis is restricted to the free-decaying regions of the signal, where the reverberation effect dominates, yielding a more accurate RT estimate at a reduced computational cost. A spectral decomposition is performed on the reverberant signal and partial RT estimates are determined in all signal subbands, providing more data to the statistical-analysis stage of the algorithm, which yields the final RT estimate. Algorithm performance is assessed using two distinct speech databases, achieving 91% and 97% correlation with the RTs measured by a standard nonblind method, indicating that the proposed method blindly estimates the RT in a reliable and consistent manner.     

A central spectrum theory of binaural processing. The binaural edge pitch revisited

Recently, Klein and Hartmann [J. Acoust. Soc. Am. 70, 51-61 (1981)] investigated a new dichotic pitch, called the binaural edge pitch (BEP). They used computer-generated periodic noise signals to generate BEP. In the study presented here, the BEP is investigated in order to evaluate the predictions of the central spectrum theory with regard to this stimulus. Pitch-matching experiments using a nonperiodic BEP stimulus, produced by means of a modulation technique, led to the conclusion that the strongest pitch sensation in the BEP has the character of a weak fluctuating pure tones in noise, which corresponds to a frequency, equal or almost equal to the frequency of the phase transition. This result fits in with the predictions of the central spectrum theory, which, for instance, does not need the assumption of central lateral inhibition for explaining this pitch. Furthermore, it is shown that this theory can also predict the results obtained in lateralization measurements and BMLD measurements using BEP stimuli as well as related stimuli. The results are compared with the data obtained by Klein and Hartmann.     

The intelligibility of speech in elementary school classrooms

This is the second of two papers describing the results of acoustical measurements and speech intelligibility tests in elementary school classrooms. The intelligibility tests were performed in 41 classrooms in 12 different schools evenly divided among grades 1, 3, and 6 students (nominally 6, 8, and 11 year olds). Speech intelligibility tests were carried out on classes of students seated at their own desks in their regular classrooms. Mean intelligibility scores were significantly related to signal-to-noise ratios and to the grade of the students. While the results are different than those from some previous laboratory studies that included less realistic conditions, they agree with previous in-classroom experiments. The results indicate that +15 dB signal-to-noise ratio is not adequate for the youngest children. By combining the speech intelligibility test results with measurements of speech and noise levels during actual teaching situations, estimates of the fraction of students experiencing near-ideal acoustical conditions were made. The results are used as a basis for estimating ideal acoustical criteria for elementary school classrooms.     

Multichannel time-reversal processing for acoustic communications in a highly reverberant environment

The development of time-reversal (T/R) communication systems is a recent signal processing research area dominated by applying T/R techniques to communicate in hostile environments. The fundamental concept is based on time-reversing the impulse response or Green's function characterizing the uncertain communications channel to mitigate deleterious dispersion and multipath effects. In this paper, we extend point-to-point to array-to-point communications by first establishing the basic theory to define and solve the underlying multichannel communications problem and then developing various realizations of the resulting T/R receivers. We show that not only do these receivers perform well in a hostile environment, but they also can be implemented with a "1 bit" analog-to-digital converter design structure. We validate these results by performing proof-of-principle acoustic communications experiments in air. It is shown that the resulting T/R receivers are capable of extracting the transmitted coded sequence from noisy microphone array measurements with zero-bit error.     

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。