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。      

Effects of in-the-ear microphone directionality on sound direction identification

As advanced signal processing algorithms have been proposed to enhance hearing protective device (HPD) performance, it is important to determine how directional microphones might affect the localization ability of users and whether they might cause safety hazards. The effect of in-the-ear microphone directivity was assessed by measuring sound source identification of speech in the horizontal plane. Recordings of speech in quiet and in noise were made with Knowles Electronic Manikin for Acoustic Research wearing bilateral in-the-ear hearing aids with microphones having adjustable directivity (omnidirectional, cardioid, hypercardioid, supercardioid). Signals were generated from 16 locations in a circular array. Sound direction identification performance of eight normal hearing listeners and eight hearing-impaired listeners revealed that directional microphones did not degrade localization performance and actually reduced the front-back and lateral localization errors made when listening through omnidirectional microphones. The summed rms speech level for the signals entering the two ears appear to serve as a cue for making front-back discriminations when using directional microphones in the experimental setting. The results of this study show that the use of matched directional microphones when worn bilaterally do not have a negative effect on the ability to localize speech in the horizontal plane and may thus be useful in HPD design.     

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.     

Multi-microphone adaptive noise reduction strategies for coordinated stimulation in bilateral cochlear implant devices

Bilateral cochlear implant (BI-CI) recipients achieve high word recognition scores in quiet listening conditions. Still, there is a substantial drop in speech recognition performance when there is reverberation and more than one interferers. BI-CI users utilize information from just two directional microphones placed on opposite sides of the head in a so-called independent stimulation mode. To enhance the ability of BI-CI users to communicate in noise, the use of two computationally inexpensive multi-microphone adaptive noise reduction strategies exploiting information simultaneously collected by the microphones associated with two behind-the-ear (BTE) processors (one per ear) is proposed. To this end, as many as four microphones are employed (two omni-directional and two directional) in each of the two BTE processors (one per ear). In the proposed two-microphone binaural strategies, all four microphones (two behind each ear) are being used in a coordinated stimulation mode. The hypothesis is that such strategies combine spatial information from all microphones to form a better representation of the target than that made available with only a single input. Speech intelligibility is assessed in BI-CI listeners using IEEE sentences corrupted by up to three steady speech-shaped noise sources. Results indicate that multi-microphone strategies improve speech understanding in single- and multi-noise source scenarios.     

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.     

Localization of multiple acoustic sources with small arrays using a coherence test

Direction finding of more sources than sensors is appealing in situations with small sensor arrays. Potential applications include surveillance, teleconferencing, and auditory scene analysis for hearing aids. A new technique for time-frequency-sparse sources, such as speech and vehicle sounds, uses a coherence test to identify low-rank time-frequency bins. These low-rank bins are processed in one of two ways: (1) narrowband spatial spectrum estimation at each bin followed by summation of directional spectra across time and frequency or (2) clustering low-rank covariance matrices, averaging covariance matrices within clusters, and narrowband spatial spectrum estimation of each cluster. Experimental results with omnidirectional microphones and colocated directional microphones demonstrate the algorithm's ability to localize 3-5 simultaneous speech sources over 4 s with 2-3 microphones to less than 1 degree of error, and the ability to localize simultaneously two moving military vehicles and small arms gunfire.     

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.     

Experimental results for multichannel feedforward ANC with noninvasive system identification

This paper presents experimental validation of a class of algorithms designed to enable active noise control (ANC) to function in environments when transfer functions change significantly over time. The experimental results presented are for broadband, local quieting in a diffuse field using a multichannel ANC system. The reverberant enclosure is an ordinary room, measuring approximately 1.4 x 2.4 x 2.4 m3 and containing a seated occupant, with six microphones defining the quiet zone near the occupant's ears. The control system uses a single reference signal and two error channels to drive four secondary sources. Using an ideal reference sensor, reduction in sound pressure level is obtained at the quiet-zone microphones averaged over the frequency range 50 to 1000 Hz with an occupant seated in the room. Two main results are presented: first for an adaptive cancelling algorithm that uses static system models, and second for the same algorithm joined with a noninvasive real-time system identification algorithm. In the first case better than 23 dB of performance is obtained if the occupant remains still through calibration and testing. In the second case, approximately 18 dB is obtained at the error microphones regardless of the motion of the occupant.     

Variable step-size subband backward BSS algorithms for speech quality enhancement

This paper addresses the problem of the speech quality improvement using adaptive filtering algorithms. Recently in Djendi and Bendoumia (2014) [1], we have proposed a new two-channel backward algorithm for noise reduction and speech intelligibility enhancement. The main drawback of proposed two-channel subband algorithm is its poor performance when the number of subband is high. This inconvenience is well seen in the steady state regime values. The source of this problem is the fixed step-sizes of the cross-adaptive filtering algorithms that distort the speech signal when they are selected high and degrade the convergence speed behaviours when they are selected small. In this paper, we propose four modifications of this algorithm which allow improving both the convergence speed and the steady state values even in very noisy condition and a high number of subbands. To confirm the good performance of the four proposed variable-step-size SBBSS algorithms, we have carried out several simulations in various noisy environments. In these simulations, we have evaluated objective and subjective criteria as the system mismatch, the cepstral distance, the output signal-to-noise-ratio, and the mean opinion score (MOS) method to compare the four proposed variables step-size versions of the SBBSS algorithm with their original versions and with the two-channel fullband backward (2CFB) least mean square algorithm.     

Speech intelligibility and spatial release from masking in young children

Children between the ages of 4 and 7 and adults were tested in free field on speech intelligibility using a four-alternative forced choice paradigm with spondees. Target speech was presented from front (0 degrees); speech or modulated speech-shaped-noise competitors were either in front or on the right (90 degrees). Speech reception thresholds were measured adaptively using a three-down/one-up algorithm. The primary difference between children and adults was seen in elevated thresholds in children in quiet and in all masked conditions. For both age groups, masking was greater with the speech-noise versus speech competitor and with two versus one competitor(s). Masking was also greater when the competitors were located in front compared with the right. The amount of masking did not differ across the two age groups. Spatial release from masking was similar in the two age groups, except for in the one-speech condition, when it was greater in children than adults. These findings suggest that, similar to adults, young children are able to utilize spatial and/or head shadow cues to segregate sounds in noisy environments. The potential utility of the measures used here for studying hearing-impaired children is also discussed.     

The spatial unmasking of speech: evidence for better-ear listening

Speech reception thresholds (SRTs) were measured for target speech presented concurrently with interfering speech (spoken by a different speaker). In experiment 1, the target and interferer were divided spectrally into high- and low-frequency bands and presented over headphones in three conditions: monaural, dichotic (target and interferer to different ears), and swapped (the low-frequency target band and the high-frequency interferer band were presented to one ear, while the high-frequency target band and the low-frequency interferer band were presented to the other ear). SRTs were highest in the monaural condition and lowest in the dichotic condition; SRTs in the swapped condition were intermediate. In experiment 2, two new conditions were devised such that one target band was presented in isolation to one ear while the other band was presented at the other ear with the interferer. The pattern of SRTs observed in experiment 2 suggests that performance in the swapped condition reflects the intelligibility of the target frequency bands at just one ear; the auditory system appears unable to exploit advantageous target-to-interferer ratios at different ears when segregating target speech from a competing speech interferer.     

Active stereo sound localization

Estimating the direction of arrival of sound in three-dimensional space is typically performed by generalized time-delay processing on a set of signals from a fixed array of omnidirectional microphones. This requires specialized multichannel A/D hardware, and careful arrangement of the microphones into an array. This work is motivated by the desire to instead only use standard two-channel audio A/D hardware and portable equipment. To estimate direction of arrival of persistent sound, the position of the microphones is made variable by mounting them on one or more computer-controlled pan-and-tilt units. In this paper, we describe the signal processing and control algorithm of a device with two omnidirectional microphones on a fixed baseline and two rotational degrees of freedom. Experimental results with real data are reported with both impulsive and speech sounds in an untreated, normally reverberant indoor environment.     

The effect of multimicrophone noise reduction systems on sound source localization by users of binaural hearing aids

This paper evaluates the influence of three multimicrophone noise reduction algorithms on the ability to localize sound sources. Two recently developed noise reduction techniques for binaural hearing aids were evaluated, namely, the binaural multichannel Wiener filter (MWF) and the binaural multichannel Wiener filter with partial noise estimate (MWF-N), together with a dual-monaural adaptive directional microphone (ADM), which is a widely used noise reduction approach in commercial hearing aids. The influence of the different algorithms on perceived sound source localization and their noise reduction performance was evaluated. It is shown that noise reduction algorithms can have a large influence on localization and that (a) the ADM only preserves localization in the forward direction over azimuths where limited or no noise reduction is obtained; (b) the MWF preserves localization of the target speech component but may distort localization of the noise component. The latter is dependent on signal-to-noise ratio and masking effects; (c) the MWF-N enables correct localization of both the speech and the noise components; (d) the statistical Wiener filter approach introduces a better combination of sound source localization and noise reduction performance than the ADM approach.     

Beamformer performance with acoustic vector sensors in air

For some time, compact acoustic vector sensors (AVSs) capable of sensing particle velocity in three orthogonal directions have been used in underwater acoustic sensing applications. Potential advantages of using AVSs in air include substantial noise reduction with a very small aperture and few channels. For this study, a four-microphone array approximating a small (1 cm3) AVS in air was constructed using three gradient microphones and one omnidirectional microphone. This study evaluates the signal extraction performance of one nonadaptive and four adaptive beamforming algorithms. Test signals, consisting of two to five speech sources, were processed with each algorithm, and the signal extraction performance was quantified by calculating the signal-to-noise ratio (SNR) of the output. For a three-microphone array, robust and nonrobust versions of a frequency-domain minimum-variance (FMV) distortionless-response beamformer produced SNR improvements of 11 to 14 dB, and a generalized sidelobe canceller (GSC) produced improvements of 5.5 to 8.5 dB. In comparison, a two-microphone omnidirectional array with a spacing of 15 cm yielded slightly lower SNR improvements for similar multi-interferer speech signals.     

Speech masking. II: Simultaneous masking thresholds under "naturalistic" listening conditions

This article investigates the role of listening conditions in determining thresholds for probe tones masked by natural speech. These thresholds are of interest because they are a sensitive probe of the activity profile, or spectrum, of sounds such as speech in the auditory system. Most human performance tests are carried out under highly artificial listening conditions, which may not reflect how people listen to speech in common listening environments. In this study, reference conditions (similar to minimal uncertainty listening conditions used in many performance tests) were compared to a "naturalistic" listening condition and to another, intermediate, condition. In the naturalistic listening condition, listeners did not know the frequency or the position of probe tones; additionally, they were required to attend to the semantic content of sentences. In the reference condition, listeners knew the frequency and position of probe tones masked by single syllables. Average thresholds were elevated by 4 dB in the naturalistic listening condition with respect to the reference condition, and thresholds tended to be elevated more for higher-frequency probe tones. The results provide previously unknown information about the resolution of speech sounds in the auditory system during speech comprehension.     

Horizontal localization with bilateral hearing aids: without is better than with

This paper studies the effect of bilateral hearing aids on directional hearing in the frontal horizontal plane. Localization tests evaluated bilateral hearing aid users using different stimuli and different noise scenarios. Normal hearing subjects were used as a reference. The main research questions raised in this paper are: (i) How do bilateral hearing aid users perform on a localization task, relative to normal hearing subjects? (ii) Do bilateral hearing aids preserve localization cues, and (iii) Is there an influence of state of the art noise reduction algorithms, more in particular an adaptive directional microphone configuration, on localization performance? The hearing aid users were tested without and with their hearing aids, using both a standard omnidirectional microphone configuration and an adaptive directional microphone configuration. The following main conclusions are drawn. (i) Bilateral hearing aid users perform worse than normal hearing subjects in a localization task, although more than one-half of the subjects reach normal hearing performance when tested unaided. For both groups, localization performance drops significantly when acoustical scenarios become more complex. (ii) Bilateral, i.e., independently operating hearing aids do not preserve localization cues. (iii) Overall, adaptive directional noise reduction can have an additional and significant negative impact on localization performance.     

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.     

Auditory and nonauditory factors affecting speech reception in noise by older listeners

Speech reception thresholds (SRTs) for sentences were determined in stationary and modulated background noise for two age-matched groups of normal-hearing (N = 13) and hearing-impaired listeners (N = 21). Correlations were studied between the SRT in noise and measures of auditory and nonauditory performance, after which stepwise regression analyses were performed within both groups separately. Auditory measures included the pure-tone audiogram and tests of spectral and temporal acuity. Nonauditory factors were assessed by measuring the text reception threshold (TRT), a visual analogue of the SRT, in which partially masked sentences were adaptively presented. Results indicate that, for the normal-hearing group, the variance in speech reception is mainly associated with nonauditory factors, both in stationary and in modulated noise. For the hearing-impaired group, speech reception in stationary noise is mainly related to the audiogram, even when audibility effects are accounted for. In modulated noise, both auditory (temporal acuity) and nonauditory factors (TRT) contribute to explaining interindividual differences in speech reception. Age was not a significant factor in the results. It is concluded that, under some conditions, nonauditory factors are relevant for the perception of speech in noise. Further evaluation of nonauditory factors might enable adapting the expectations from auditory rehabilitation in clinical settings.     

Evaluation of Blind Source Separation for different algorithms based on second order statistics and different spatial configurations of directional microphones

The present study is concerned with the convolutive Blind Source Separation (BSS) of sound sources that leads to a significant speech intelligibility enhancement. Two experiments were conducted. In the first experiment two different algorithms of convolutive BSS were compared. Both methods are based on second order statistics since such approach is simple and gives satisfactory performance. The data resulted from this experiment suggested that with different approaches, different speech intelligibility improvement could be obtained. In the second experiment the influence of the spatial configuration of the cardioid microphones on the BSS performance was measured. It was revealed that the best separation for a considered spatial configuration can be obtained when microphones are directed alternately.