Comparing the effects of reverberation and of noise on speech recognition in simulated electric-acoustic listening

Cochlear implant users report difficulty understanding speech in both noisy and reverberant environments. Electric-acoustic stimulation (EAS) is known to improve speech intelligibility in noise. However, little is known about the potential benefits of EAS in reverberation, or about how such benefits relate to those observed in noise. The present study used EAS simulations to examine these questions. Sentences were convolved with impulse responses from a model of a room whose estimated reverberation times were varied from 0 to 1 sec. These reverberated stimuli were then vocoded to simulate electric stimulation, or presented as a combination of vocoder plus low-pass filtered speech to simulate EAS. Monaural sentence recognition scores were measured in two conditions: reverberated speech and speech in a reverberated noise. The long-term spectrum and amplitude modulations of the noise were equated to the reverberant energy, allowing a comparison of the effects of the interferer (speech vs noise). Results indicate that, at least in simulation, (1) EAS provides significant benefit in reverberation; (2) the benefits of EAS in reverberation may be underestimated by those in a comparable noise; and (3) the EAS benefit in reverberation likely arises from partially preserved cues in this background accessible via the low-frequency acoustic component.     

Speech intelligibility in cochlear implant simulations: Effects of carrier type, interfering noise, and subject experience

Channel vocoders using either tone or band-limited noise carriers have been used in experiments to simulate cochlear implant processing in normal-hearing listeners. Previous results from these experiments have suggested that the two vocoder types produce speech of nearly equal intelligibility in quiet conditions. The purpose of this study was to further compare the performance of tone and noise-band vocoders in both quiet and noisy listening conditions. In each of four experiments, normal-hearing subjects were better able to identify tone-vocoded sentences and vowel-consonant-vowel syllables than noise-vocoded sentences and syllables, both in quiet and in the presence of either speech-spectrum noise or two-talker babble. An analysis of consonant confusions for listening in both quiet and speech-spectrum noise revealed significantly different error patterns that were related to each vocoder's ability to produce tone or noise output that accurately reflected the consonant's manner of articulation. Subject experience was also shown to influence intelligibility. Simulations using a computational model of modulation detection suggest that the noise vocoder's disadvantage is in part due to the intrinsic temporal fluctuations of its carriers, which can interfere with temporal fluctuations that convey speech recognition cues.     

Conversational quality evaluation of artificial bandwidth extension of telephone speech

Artificial bandwidth extension methods have been developed to improve the quality and intelligibility of narrowband telephone speech and to reduce the difference with wideband speech. Such methods have commonly been evaluated with objective measures or subjective listening-only tests, but conversational evaluations have been rare. This article presents a conversational evaluation of two methods for the artificial bandwidth extension of telephone speech. Bandwidth-extended narrowband speech is compared with narrowband and wideband speech in a test setting including a simulated telephone connection, realistic conversation tasks, and various background noise conditions. The responses of the subjects indicate that speech processed with one of the methods is preferred to narrowband speech in noise, but wideband speech is superior to both narrowband and bandwidth-extended speech. Bandwidth extension was found to be beneficial for telephone conversation in noisy listening conditions.     

Comparative intelligibility investigation of single-channel noise-reduction algorithms for Chinese, Japanese, and English

A large number of single-channel noise-reduction algorithms have been proposed based largely on mathematical principles. Most of these algorithms, however, have been evaluated with English speech. Given the different perceptual cues used by native listeners of different languages including tonal languages, it is of interest to examine whether there are any language effects when the same noise-reduction algorithm is used to process noisy speech in different languages. A comparative evaluation and investigation is taken in this study of various single-channel noise-reduction algorithms applied to noisy speech taken from three languages: Chinese, Japanese, and English. Clean speech signals (Chinese words and Japanese words) were first corrupted by three types of noise at two signal-to-noise ratios and then processed by five single-channel noise-reduction algorithms. The processed signals were finally presented to normal-hearing listeners for recognition. Intelligibility evaluation showed that the majority of noise-reduction algorithms did not improve speech intelligibility. Consistent with a previous study with the English language, the Wiener filtering algorithm produced small, but statistically significant, improvements in intelligibility for car and white noise conditions. Significant differences between the performances of noise-reduction algorithms across the three languages were observed.     

Evaluation of speech transmission in open public spaces affected by combined noises

In the present study, the effects of interference from combined noises on speech transmission were investigated in a simulated open public space. Sound fields for dominant noises were predicted using a typical urban square model surrounded by buildings. Then road traffic noise and two types of construction noises, corresponding to stationary and impulsive noises, were selected as background noises. Listening tests were performed on a group of adults, and the quality of speech transmission was evaluated using listening difficulty as well as intelligibility scores. During the listening tests, two factors that affect speech transmission performance were considered: (1) temporal characteristics of construction noise (stationary or impulsive) and (2) the levels of the construction and road traffic noises. The results indicated that word intelligibility scores and listening difficulty ratings were affected by the temporal characteristics of construction noise due to fluctuations in the background noise level. It was also observed that listening difficulty is unable to describe the speech transmission in noisy open public spaces showing larger variation than did word intelligibility scores.     

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。      

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.     

Quantifying the intelligibility of speech in noise for non-native listeners

When listening to languages learned at a later age, speech intelligibility is generally lower than when listening to one's native language. The main purpose of this study is to quantify speech intelligibility in noise for specific populations of non-native listeners, only broadly addressing the underlying perceptual and linguistic processing. An easy method is sought to extend these quantitative findings to other listener populations. Dutch subjects listening to Germans and English speech, ranging from reasonable to excellent proficiency in these languages, were found to require a 1-7 dB better speech-to-noise ratio to obtain 50% sentence intelligibility than native listeners. Also, the psychometric function for sentence recognition in noise was found to be shallower for non-native than for native listeners (worst-case slope around the 50% point of 7.5%/dB, compared to 12.6%/dB for native listeners). Differences between native and non-native speech intelligibility are largely predicted by linguistic entropy estimates as derived from a letter guessing task. Less effective use of context effects (especially semantic redundancy) explains the reduced speech intelligibility for non-native listeners. While measuring speech intelligibility for many different populations of listeners (languages, linguistic experience) may be prohibitively time consuming, obtaining predictions of non-native intelligibility from linguistic entropy may help to extend the results of this study to other listener populations.     

Effects of simulated cochlear-implant processing on speech reception in fluctuating maskers

This study investigated the effects of simulated cochlear-implant processing on speech reception in a variety of complex masking situations. Speech recognition was measured as a function of target-to-masker ratio, processing condition (4, 8, 24 channels, and unprocessed) and masker type (speech-shaped noise, amplitude-modulated speech-shaped noise, single male talker, and single female talker). The results showed that simulated implant processing was more detrimental to speech reception in fluctuating interference than in steady-state noise. Performance in the 24-channel processing condition was substantially poorer than in the unprocessed condition, despite the comparable representation of the spectral envelope. The detrimental effects of simulated implant processing in fluctuating maskers, even with large numbers of channels, may be due to the reduction in the pitch cues used in sound source segregation, which are normally carried by the peripherally resolved low-frequency harmonics and the temporal fine structure. The results suggest that using steady-state noise to test speech intelligibility may underestimate the difficulties experienced by cochlear-implant users in fluctuating acoustic backgrounds.     

Predicting speech intelligibility based on the signal-to-noise envelope power ratio after modulation-frequency selective processing

A model for predicting the intelligibility of processed noisy speech is proposed. The speech-based envelope power spectrum model has a similar structure as the model of Ewert and Dau [(2000). J. Acoust. Soc. Am. 108, 1181-1196], developed to account for modulation detection and masking data. The model estimates the speech-to-noise envelope power ratio, SNR(env), at the output of a modulation filterbank and relates this metric to speech intelligibility using the concept of an ideal observer. Predictions were compared to data on the intelligibility of speech presented in stationary speech-shaped noise. The model was further tested in conditions with noisy speech subjected to reverberation and spectral subtraction. Good agreement between predictions and data was found in all cases. For spectral subtraction, an analysis of the model's internal representation of the stimuli revealed that the predicted decrease of intelligibility was caused by the estimated noise envelope power exceeding that of the speech. The classical concept of the speech transmission index fails in this condition. The results strongly suggest that the signal-to-noise ratio at the output of a modulation frequency selective process provides a key measure of speech intelligibility.     

活动场所的语言清晰度和鸡尾酒会效应

本文讨论人们活动场所的语言清晰度，鸡尾酒会效应和它对语言清晰度的影响，对会议厅，剧院，教室，侯机大厅，侯车厅，体育馆，溜冰场等处的测量和清晰度试验，建立了主观评价和清晰度得分关系并给出一些这类场所不引起鸡尾酒会效应的允噪声级阈值，最后讨论了鸡尾酒会效应的计算机模型。     

Perceptually optimized gain function for cochlear implant signal-to-noise ratio based noise reduction

Noise reduction in cochlear implants has achieved significant speech perception improvements through spectral subtraction and signal-to-noise ratio based noise reduction techniques. Current methods use gain functions derived through mathematical optimization or motivated by normal listening psychoacoustic experiments. Although these gain functions have been able to improve speech perception, recent studies have indicated that they are not optimal for cochlear implant noise reduction. This study systematically investigates cochlear implant recipients' speech perception and listening preference of noise reduction with a range of gain functions. Results suggest an advantageous gain function and show that gain functions currently used for noise reduction are not optimal for cochlear implant recipients. Using the cochlear implant optimised gain function, a 27% improvement over the current advanced combination encoder (ACE) stimulation strategy in speech weighted noise and a 7% improvement over current noise reduction strategies were observed in babble noise conditions. The optimized gain function was also most preferred by cochlear implant recipients. The CI specific gain function derived from this study can be easily incorporated into existing noise reduction strategies, to further improve listening performance for CI recipients in challenging environments.     

A comparative intelligibility study of single-microphone noise reduction algorithms

The evaluation of intelligibility of noise reduction algorithms is reported. IEEE sentences and consonants were corrupted by four types of noise including babble, car, street and train at two signal-to-noise ratio levels (0 and 5 dB), and then processed by eight speech enhancement methods encompassing four classes of algorithms: spectral subtractive, sub-space, statistical model based and Wiener-type algorithms. The enhanced speech was presented to normal-hearing listeners for identification. With the exception of a single noise condition, no algorithm produced significant improvements in speech intelligibility. Information transmission analysis of the consonant confusion matrices indicated that no algorithm improved significantly the place feature score, significantly, which is critically important for speech recognition. The algorithms which were found in previous studies to perform the best in terms of overall quality, were not the same algorithms that performed the best in terms of speech intelligibility. The subspace algorithm, for instance, was previously found to perform the worst in terms of overall quality, but performed well in the present study in terms of preserving speech intelligibility. Overall, the analysis of consonant confusion matrices suggests that in order for noise reduction algorithms to improve speech intelligibility, they need to improve the place and manner feature scores.     

Improvements in intelligibility of noisy reverberant speech using a binaural subband adaptive noise-cancellation processing scheme.

This article reports on the performance of an adaptive subband noise cancellation scheme, which performs binaural preprocessing of speech signals for a hearing-aid application. The multi-microphone subband adaptive (MMSBA) signal processing scheme uses the least mean squares (LMS) algorithm in frequency-limited subbands. The use of subbands enables a diverse processing mechanism to be employed, splitting the two-channel wide-band signal into smaller frequency-limited subbands, which can be processed according to their individual signal characteristics. The frequency delimiting used a linear- or cochlear-spaced subband distribution. The effect of the processing scheme on speech intelligibility was assessed in a trial involving 15 hearing-impaired volunteers with moderate sensorineural hearing loss. The acoustic material consisted of speech and speech-shaped noise signals, generated using simulated and real-room acoustic environments, at signal-to-noise ratios (SNRs) in the range -6 to +3 dB. The results show that the MMSBA scheme delivered average speech intelligibility improvements of 11.5%, with a maximum of 37.25%, in noisy reverberant conditions. There was no significant reduction in mean speech intelligibility due to processing, in any of the test conditions.     

Effects of noise on speech production: acoustic and perceptual analyses

Acoustical analyses were carried out on a set of utterances produced by two male speakers talking in quiet and in 80, 90, and 100 dB SPL of masking noise. In addition to replicating previous studies demonstrating increases in amplitude, duration, and vocal pitch while talking in noise, these analyses also found reliable differences in the formant frequencies and short-term spectra of vowels. Perceptual experiments were also conducted to assess the intelligibility of utterances produced in quiet and in noise when they were presented at equal S/N ratios for identification. In each experiment, utterances originally produced in noise were found to be more intelligible than utterances produced in the quiet. The results of the acoustic analyses showed clear and consistent differences in the acoustic-phonetic characteristics of speech produced in quiet versus noisy environments. Moreover, these accounts differences produced reliable effects on intelligibility. The findings are discussed in terms of: (1) the nature of the acoustic changes that taken place when speakers produce speech under adverse conditions such as noise, psychological stress, or high cognitive load: (2) the role of training and feedback in controlling and modifying a talker's speech to improve performance of current speech recognizers; and (3) the development of robust algorithms for recognition of speech in noise.     

Contribution of frequency modulation to speech recognition in noise

Cochlear implants allow most patients with profound deafness to successfully communicate under optimal listening conditions. However, the amplitude modulation (AM) information provided by most implants is not sufficient for speech recognition in realistic settings where noise is typically present. This study added slowly varying frequency modulation (FM) to the existing algorithm of an implant simulation and used competing sentences to evaluate FM contributions to speech recognition in noise. Potential FM advantage was evaluated as a function of the number of spectral bands, FM depth, FM rate, and FM band distribution. Barring floor and ceiling effects, significant improvement was observed for all bands from 1 to 32 with the additional FM cue both in quiet and noise. Performance also improved with greater FM depth and rate, which might reflect resolved sidebands under the FM condition. Having FM present in low-frequency bands was more beneficial than in high-frequency bands, and only half of the bands required the presence of FM, regardless of position, to achieve performance similar to when all bands had the FM cue. These results provide insight into the relative contributions of AM and FM to speech communication and the potential advantage of incorporating FM for cochlear implant signal processing.     

Recognition of accented English in quiet and noise by younger and older listeners

This study investigated the effects of age and hearing loss on perception of accented speech presented in quiet and noise. The relative importance of alterations in phonetic segments vs. temporal patterns in a carrier phrase with accented speech also was examined. English sentences recorded by a native English speaker and a native Spanish speaker, together with hybrid sentences that varied the native language of the speaker of the carrier phrase and the final target word of the sentence were presented to younger and older listeners with normal hearing and older listeners with hearing loss in quiet and noise. Effects of age and hearing loss were observed in both listening environments, but varied with speaker accent. All groups exhibited lower recognition performance for the final target word spoken by the accented speaker compared to that spoken by the native speaker, indicating that alterations in segmental cues due to accent play a prominent role in intelligibility. Effects of the carrier phrase were minimal. The findings indicate that recognition of accented speech, especially in noise, is a particularly challenging communication task for older people.     

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 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 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.