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.     

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.     

Rician noise reduction in magnetic resonance images using adaptive non-local mean and guided image filtering

This paper proposes a Rician noise reduction method for magnetic resonance (MR) images. The proposed method is based on adaptive non-local mean and guided image filtering techniques. In the first phase, a guidance image is obtained from the noisy image through an adaptive non-local mean filter. Sobel operators are applied to compute the strength of edges which is further used to control the spread of the kernel in non-local mean filtering. In the second phase, the noisy and the guidance images are provided to the guided image filter as input to restore the noise-free image. The improved performance of the proposed method is investigated using the simulated and real data sets of MR images. Its performance is also compared with the previously proposed state-of-the art methods. Comparative analysis demonstrates the superiority of the proposed scheme over the existing approaches.     

Localization ability with bimodal hearing aids and bilateral cochlear implants

After successful cochlear implantation in one ear, some patients continue to use a hearing aid at the contralateral ear. They report an improved reception of speech, especially in noise, as well as a better perception of music when the hearing aid and cochlear implant are used in this bimodal combination. Some individuals in this bimodal patient group also report the impression of an improved localization ability. Similar experiences are reported by the group of bilateral cochlear implantees. In this study, a survey of 11 bimodally and 4 bilaterally equipped cochlear implant users was carried out to assess localization ability. Individuals in the bimodal implant group were all provided with the same type of hearing aid in the opposite ear, and subjects in the bilateral implant group used cochlear implants of the same manufacturer on each ear. Subjects adjusted the spot of a computer-controlled laser-pointer to the perceived direction of sound incidence in the frontal horizontal plane by rotating a trackball. Two subjects of the bimodal group who had substantial residual hearing showed localization ability in the bimodal configuration, whereas using each single device only the subject with better residual hearing was able to discriminate the side of sound origin. Five other subjects with more pronounced hearing loss displayed an ability for side discrimination through the use of bimodal aids, while four of them were already able to discriminate the side with a single device. Of the bilateral cochlear implant group one subject showed localization accuracy close to that of normal hearing subjects. This subject was also able to discriminate the side of sound origin using the first implanted device alone. The other three bilaterally equipped subjects showed limited localization ability using both devices. Among them one subject demonstrated a side-discrimination ability using only the first implanted device.     

Speech-reception threshold in noise with one and two hearing aids

The binaural free-field speech-reception threshold (SRT) in 70-dBA noise was measured with conversational sentences for 24 hearing-impaired subjects without hearing aids, with a hearing aid left, right, and left plus right, respectively. The sentences were always presented in front of the listener and the interfering noise, with a spectrum equal to the long-term average spectrum of the sentences, was presented either frontally, from the right, or from the left side. For subjects with only moderate hearing loss, PTA (average air-conduction hearing level at 500, 1000, and 2000 Hz) less than 50 dB, the SRT in 70-dBA noise in both ears is determined by the signal-to-noise ratio even if only one hearing aid is used. For larger hearing losses the SRT appears to be partly determined by the absolute threshold. In conditions with a high noise level relative to the absolute threshold, in which case for both ears the SRT is determined by the signal-to-noise ratio, a second hearing aid, just as a monaural hearing aid, generally does not improve the SRT. However, in the case of a high hearing level, or a low noise level, in which a monaural hearing aid is profitable, the use of two hearing aids is even more profitable. In a separate experiment, acoustic head shadow was measured at the entrance of the ear canal and at the microphone location of a hearing aid. It appeared that, for a lateral noise source and speech frontal, the microphone position of behind-the-ear hearing aids has a negative effect on the signal-to-noise ratio of 2-3 dB.     

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.     

Speech production in noise with and without hearing protection

People working in noisy environments often complain of difficulty communicating when they wear hearing protection. It was hypothesized that part of the workers' communication difficulties stem from changes in speech production that occur when hearing protectors are worn. To address this possibility, overall and one-third-octave-band SPL measurements were obtained for 16 men and 16 women as they produced connected speech while wearing foam, flange, or no earplugs (open ears) in quiet and in pink noise at 60, 70, 80, 90, and 100 dB SPL. The attenuation and the occlusion effect produced by the earplugs were measured. The Speech Intelligibility Index (SII) was also calculated for each condition. The talkers produced lower overall speech levels, speech-to-noise ratios, and SII values, and less high-frequency speech energy, when they wore earplugs compared with the open-ear condition. Small differences in the speech measures between the talkers wearing foam and flange earplugs were observed. Overall, the results of the study indicate that talkers wearing earplugs (and consequently their listeners) are at a disadvantage when communicating in noise.     

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.     

Speech perception studies using a multichannel electrotactile speech processor, residual hearing, and lipreading

Three studies are reported on the speech perception of normally hearing and hearing-impaired adults using combinations of visual, auditory, and tactile input. In study 1, mean scores for four normally hearing subjects showed that addition of tactile information, provided through the multichannel electrotactile speech processor, to either audition alone (300-Hz low-pass-filtered speech) or lipreading plus audition resulted in significant improvements in phoneme and word discrimination scores. Information transmission analyses demonstrated the effectiveness of the tactile aid in providing cues to duration, F1 and F2 features for vowels, and manner of articulation features for consonants, especially features requiring detection and discrimination of high-frequency information. In study 2, six different cutoff frequencies were used for a low-pass-filtered auditory signal. Mean scores for vowel and consonant identification were significantly higher with the addition of tactile input to audition alone at each cutoff frequency up to 1500 Hz. The mean speechtracking rate was also significantly increased by the additional tactile input up to 1500 Hz. Study 3 examined speech discrimination of three hearing-impaired adults. Additional information available through the tactile aid was shown to improve speech discrimination scores; however, the degree of increase was inversely related to the level of residual hearing. Results indicate that the electrotactile aid may be useful for patients with little residual hearing and for the severely to profoundly hearing impaired, who could benefit from the high-frequency information presented through the tactile modality, but unavailable through hearing aids.     

基于阵列抗串扰自适应噪声抵消与谱相减相结合的语音增强

基于传声器阵的语音增强对于语音通信及语音识别具有重要意义,减小传声器阵的孔径并提高其语音增强的效果将十分有利于扩大其应用范围。为此给出一种阵列抗串扰自适应噪声抵消方法,并将其与改进的谱相减技术相结合。分析与实验表明,该方案使增强语音的信噪比有了较大提高,所需的阵列孔径很小,而且还适用于多种噪声环境,并易于实时实现。     

Speech recognition in noise with active and passive hearing protectors: a comparative study

The perceived negative influence of standard hearing protectors on communication is a common argument for not wearing them. Thus, "augmented" protectors have been developed to improve speech intelligibility. Nevertheless, their actual benefit remains a point of concern. In this paper, speech perception with active earplugs is compared to standard passive custom-made earplugs. The two types of active protectors included amplify the incoming sound with a fixed level or to a user selected fraction of the maximum safe level. For the latter type, minimal and maximal amplification are selected. To compare speech intelligibility, 20 different speech-in-noise fragments are presented to 60 normal-hearing subjects and speech recognition is scored. The background noise is selected from realistic industrial noise samples with different intensity, frequency, and temporal characteristics. Statistical analyses suggest that the protectors' performance strongly depends on the noise condition. The active protectors with minimal amplification outclass the others for the most difficult and the easiest situations, but they also limit binaural listening. In other conditions, the passive protectors clearly surpass their active counterparts. Subsequently, test fragments are analyzed acoustically to clarify the results. This provides useful information for developing prototypes, but also indicates that tests with human subjects remain essential.     

Binaural speech unmasking and localization in noise with bilateral cochlear implants using envelope and fine-timing based strategies

Four adult bilateral cochlear implant users, with good open-set sentence recognition, were tested with three different sound coding strategies for binaural speech unmasking and their ability to localize 100 and 500 Hz click trains in noise. Two of the strategies tested were envelope-based strategies that are clinically widely used. The third was a research strategy that additionally preserved fine-timing cues at low frequencies. Speech reception thresholds were determined in diotic noise for diotic and interaurally time-delayed speech using direct audio input to a bilateral research processor. Localization in noise was assessed in the free field. Overall results, for both speech and localization tests, were similar with all three strategies. None provided a binaural speech unmasking advantage due to the application of 700 micros interaural time delay to the speech signal, and localization results showed similar response patterns across strategies that were well accounted for by the use of broadband interaural level cues. The data from both experiments combined indicate that, in contrast to normal hearing, timing cues available from natural head-width delays do not offer binaural advantages with present methods of electrical stimulation, even when fine-timing cues are explicitly coded.     

Noise reduction of speech signals using time-varying and multi-band adaptive gain control for smart digital hearing protectors

In this paper, a single-channel speech enhancement algorithm based on non-linear and multi-band Adaptive Gain Control (AGC) is proposed. The algorithm requires neither Signal-to-Noise Ratio (SNR) nor noise parameters estimation. It reduces the background noise in the temporal domain rather than the spectral domain using a non-linear and automatically adjustable gain function for multi-band AGC. The gain function varies in time and is deduced from the temporal envelope of each frequency band to highly compress the frequency regions where noise is present and lightly compress the frequency regions where speech is present. Objective evaluation using the PESQ (Perceptual Evaluation of Speech Quality) metric shows that the proposed algorithm performs better than three benchmarks, namely: the spectral subtraction, the Wiener filter based on a priori SNR estimation and a band-pass modulation filtering algorithm. In addition, blind subjective tests show that the proposed algorithm introduces less musical noise compared to the benchmark algorithms and was preferred 78.8% of the time in terms of signal quality. The proposed algorithm is implemented in a miniature low power digital signal processor to validate its feasibility and complexity for smart hearing protection in noisy environments.