利用普通微机处理听觉诱发电位数据

利用普通微机处理听觉诱发电位数据赵宁军，李克勇目前，电反应测听国内已逐渐普及，耳蜗电图（ＥＣｏｃｈＧ）和听性脑干反应（ＡＢＲ）已在临床上广泛使用，有时需对ＥＣｏｃｈＧ和ＡＢＲ进行数据处理。目前多数医院尚无力购买专门的数据处理仪系统，仅能购买功能较低的...     

电刺激诱发中潜伏期反应在人工耳蜗植入中的应用

听觉诱发电位（auditory evoked potential，AEP）是声刺激后由耳蜗至听皮层的听觉系统中不同平面所诱发出的一系列电活动，包括听性脑干反应（audi-tory brain-stem response，ABR）、中潜伏期反应（mid-dle latency response，MLR）及慢皮层反应等。随着人工耳蜗植入（cochlear implant，CI）技术的逐渐发展和成熟，近40年产生了一类以电刺激为诱发形式的听觉诱发电位。这成为了CI患者，尤其是对于幼儿患者的一种重要评估手段。同时，该类技术亦在神经可塑性的研究等方面发挥了重要作用。电刺激听觉诱发电位主要包括电刺激听性脑干反应（electrically evoked auditory brain-stem response，EABR）、电刺激中潜伏期反应（electrically evoked middle latency re-sponse，EMLR）、电刺激长潜伏期反应（electrically evoked late latency response，EALR）以及认知诱发电位（cognitive evoked potential）等。由于EMLR潜伏期相对较长，可反映听觉通路中丘脑以至听皮层等高级中枢的情况，相比其他时长诱发电位具有其独特的优势，本文将对其发展及研究现状进行介绍。     

几种常用听觉诱发反应测听的比较

听觉诱发反应包括听觉诱发电位和耳声发射。通常所述的电反应测听(electric response audiometry, ERA)实际上是建立在一系列听觉诱发电位(auditory evoked potentials, AEP)检测技术基础上的客观方法。 20世纪70年代末我国开始陆续引进ERA,目前已经得到了广泛的应用,积累了丰富的经验,为临床鉴别诊断提供了重要的依据。     

听神经病患者80Hz听觉稳态诱发电位研究

目的:探讨听神经病患者80 Hz听觉稳态诱发电位的特点及与听性脑干诱发电位V波的相关性.方法:以1000 Hz短音95 dBnHL作声刺激,比较听神经病组和正常对照组80 Hz听觉稳态诱发电位及短音听性脑干反应(ABR)的波形差异.结果:听神经病组两种听觉诱发电位波形的同时引出率为85.4%且波幅低或极低,ABR的Ⅰ、Ⅲ波消失;14.6%两种听觉诱发电位波形同时消失.而正常对照组两种波形的同时引出率为100.0%.结论:80 Hz听觉稳态诱发电位与听性脑干反应的V波有明显相关性.     

电反应测听技术在临床上的应用

多年来,测量听觉系统的声刺激诱发电位已成为动物实验的一种手段。但是由于人体听觉系统的声刺激诱发电位太小,所以除手术情况下,到目前为止这些技术尚未能应用于人体。小型计算机的发展使得从背景电干扰中引出这些微小电位成为可能,乃促使电反应测听技术的临床应用得以实现。四年前作者就打算对电反应测听技术的门诊应用进行评价。本文提出自那时以来300例受试     

问题与解答（续5）

问题与解答（续５）３８电反应测听是否即为听性脑干反应？电反应测听（ＥｌｅｃｔｒｉｃａｌＲｅｓｐｏｎｓｅｓＡｕｄｉｏｍｅｔｒｙ，ＥＲＡ）是用声刺激使听觉系统发生生物电变化过程的测听法，是测试听性诱发电位（Ａｕｄｉｏｔｏｒｙｅｖｏｋｅｄｐｏｔｅｎｔｉａｌ...     

皮层听觉诱发电位在人工耳蜗植入中的应用

皮层听觉诱发电位（cortical auditory evoked potential, CAEP）是一种非侵入性脑电测量方式,能够通过波形的阈值、潜伏期与振幅评估听觉皮层的功能。近年来,CAEP的测试方式逐渐成熟,应用更加广泛。在人工耳蜗植入患者中,CAEP能够用于术前、术后听觉及言语能力、双耳听觉、音乐感知以及听觉可塑性的评估。相比较听性脑干反应（auditory brainstem response, ABR）、中潜伏期反应（middle latency response,MLR）等其它听觉诱发电位而言,CAEP潜伏期最长,能够反映完整听觉通路的情况、评价听觉皮层等高级中枢的功能及成熟度,具有独特优势。本文就CAEP在人工耳蜗患者中的研究进展进行综述。     

自闭症谱系障碍儿童的听觉诱发电位研究进展

自闭症谱系障碍儿童通常表现出对声音刺激的异常反应,前期针对其听觉功能的研究多采用客观听觉电生理测试,主要包含听性脑干反应及一系列皮层听觉诱发电位。综合这些研究中的结果提示,与正常发育儿童相比,自闭症谱系障碍儿童存在一定程度的中枢听觉信息处理障碍。本文对此进行综述。     

中国听性脑干反应临床操作规范专家共识(2020)

听性脑干反应(ABR),又称脑干听觉诱发电位(BAEP),是一种由气导或骨导声刺激诱发,起源于内耳、听神经和听觉脑干,在头颅表面记录到的神经电活动,属于短潜伏期听觉诱发电位。ABR是听觉诱发电位中最常用的测试方法,可以评估从内耳至听觉脑干的听觉通路的完整性,由于不受受试者睡眠状态(睡眠或清醒)的影响,可用于不能配合行为测听及需要进行听觉通路病变评估者,是临床不可或缺的听力学检测手段。ABR的波形、幅值、潜伏期和反应阈值等受测试环境和测试参数影响较大,规范的ABR测试操作对于保证其测试结果的可靠性、可比性以及提高临床诊疗的准确性至关重要。我国目前尚无ABR测试临床操作规范。基于我国听力学专家的临床实践经验和研究结果,我们在广泛征求耳科医师和听力技师等相关人员意见和建议的基础上形成本共识,以期规范ABR测试临床操作流程,便于我国ABR测试工作的进一步普及和推广。另外需要指出的是ABR虽可归于"客观测听",但对其波形的识别和判读是由测试者主观分析的,评估时要注意到"判读误差"的可能性。本共识不包括自动ABR(AABR)、电刺激ABR(EABR)和术中ABR测试技术。     

短纯音ABR在听力阈值评估中的应用

当一定强度的声音刺激听觉器官时,听觉系统就会产生一系列的电活动,称为听觉诱发电位(auditory evoked poten-tials,AEP)。听性脑干反应(auditory brainstemresponse,ABR)是AEP的一种,即声音刺激后出现的短潜伏期反应。ABR在不同年龄的人群中都能记录到,并具有很好的稳定性。AB     

Analysis of auditory evoked potentials by magnitude-squared coherence

Evoked potentials are usually analyzed in the time domain (voltage versus time). The most familiar frequency-domain measure, the power spectral density function, displays power as a function of frequency but doesn't distinguish signal power from noise power. The coherence function estimates, for each frequency, the ratio of signal power to total (signal plus noise) power and, thus, indicates the degree to which system output (scalp potential) is determined by input (acoustic stimulus). Coherence ranges from 0 to 1; values above specified critical values can be accepted as demonstrating statistically significant system response. In this paper, we present coherence analysis of human scalp responses to clicks and amplitude-modulated tones. In both cases, this analytic method provides insight into the spectral character of the response (for example, assisting in specifying desirable filter characteristics). Threshold sensitivity is also improved: statistically significant responses can be detected at lower intensity by coherence analysis than by inspection of time-domain waveforms.     

ICRA noises: artificial noise signals with speech-like spectral and temporal properties for hearing instrument assessment. International Collegium for Rehabilitative Audiology

Current standards involving technical specification of hearing aids provide limited possibilities for assessing the influence of the spectral and temporal characteristics of the input signal, and these characteristics have a significant effect on the output signal of many recent types of hearing aids. This is particularly true of digital hearing instruments, which typically include non-linear amplification in multiple channels. Furthermore, these instruments often incorporate additional non-linear functions such as "noise reduction" and "feedback cancellation". The output signal produced by a non-linear hearing instrument relates to the characteristics of the input signal in a complex manner. Therefore, the choice of input signal significantly influences the outcome of any acoustic or psychophysical assessment of a non-linear hearing instrument. For this reason, the International Collegium for Rehabilitative Audiology (ICRA) has introduced a collection of noise signals that can be used for hearing aid testing (including real-ear measurements) and psychophysical evaluation. This paper describes the design criteria, the realisation process, and the final selection of nine test signals on a CD. Also, the spectral and temporal characteristics of these signals are documented. The ICRA noises provide a well-specified set of speech-like noises with spectra shaped according to gender and vocal effort, and with different amounts of speech modulation simulating one or more speakers. These noises can be applied as well-specified background noise in psychophysical experiments. They can also serve as test signals for the evaluation of digital hearing aids with noise reduction. It is demonstrated that the ICRA noises show the effectiveness of the noise reduction schemes. Based on these initial measurements, some initial steps are proposed to develop a standard method of technical specification of noise reduction based on the modulation characteristics. For this purpose, the sensitivity of different noise reduction schemes is compared by measurements with ICRA noises with a varying ratio between unmodulated and modulated test signals: a modulated-unmodulated ratio. It can be anticipated that this information is important to understand the differences between the different implementations of noise reduction schemes in different hearing aid models and makes.     

瞬态诱发耳声发射信号数字滤波及相关函数分析

用数字滤波及相关函数处理技术对１０名（２０耳）学龄前儿童短声诱发的ＴＥＯＡＥ信号进行二次处理。结果表明：通过数字滤波及相关处理后的ＴＥＯＡＥ信号比原始记录的ＴＥＯＡＥ信号波形更加清晰，易于辨认，有效地提高了ＴＥＯＡＥ信号的信噪比及识别率。     

Input stimuli for obtaining frequency responses of automatic gain control hearing aids

Developing a family of frequency response curves for AGC types of hearing instruments using swept pure tones at varying input levels often produces erroneous results. This problem is caused by exceeding the threshold for activating the AGC circuit at some frequencies but not at other frequencies during the pure-tone sweep, thereby producing a different frequency response from that which would be obtained with a complex input signal such as speech-shaped noise. This measurement artifact may be minimized by ensuring that the threshold for activating the AGC circuit is either always exceeded or never exceeded during the development of a frequency response curve. Three input signals are compared for developing a family of frequency responses for an AGC hearing aid: (1) swept pure tone, (2) swept pure tone with bias tone added, and (3) shaped broad-band noise. The shaped broad-band noise appears to be the input signal of choice.     

Using response time to speech as a measure for listening effort

Abstract

Objective: Speech signals that do not differ in intelligibility might differ in listening effort. This study aimed to investigate the effect of background noise on response time to intelligible speech. Design: We added various amounts of stationary noise to spoken digit triplets and measured the influence of noise on the response time for both an identification and an arithmetic task: Task 1 ‘identify the final digit in a triplet’, and Task 2 ‘calculate the sum of the initial and the final digits in a triplet.’ Study sample: Twelve normal-hearing participants with a mean age of 30.6 years (range: 28–44 years). Results: Response time increased with lower (i.e. worse) signal to noise ratios for both tasks, even for signal to noise ratios with almost maximum intelligibility (close to 100%). The response time during the arithmetic task was more affected by the noise than during the identification task, but the arithmetic task demonstrated higher variance. Conclusions: The response time to digit triplets reduces significantly for increasing signal to noise ratios, even where speech intelligibility is optimal. These differences in response time might be related to listening effort and as such might be used to evaluate hearing-aid signal processing at positive SNRs.     

Noise suppression of transient-evoked otoacoustic emissions. I. A comparison with the non-linear method

A new method to record transient-evoked otoacoustic emissions (TEOAEs) is introduced. Click stimuli were presented both with and without a simultaneously presented wide-band noise burst. Subtraction of the recorded signal evoked by the noise burst plus click from the signal evoked by the click alone, cancelled the eardrum reflection components of the response and resulted in a measure of the emission. This was used to obtain the TEOAEs from 21 subjects for peak click stimulus levels of 48-66 dB SPL. The root-mean-square (RMS) level of the noise burst was set 10 dB higher than the peak click level, and resulted in suppression of the TEOAE by up to 20 dB. The TEOAE waveforms obtained by the new method were compared to those obtained with Kemp's non-linear method, and were indistinguishable in 20 of the 21 subjects. On basis of the emission spectra, they were indistinguishable in 18 out of 21 subjects. The latencies of narrow-band filtered components from the TEOAEs obtained with the two methods were also similar. This suggests that this noise-suppression method produces similar results as Kemp's non-linear method with the advantage that emission components with very short latencies can be obtained.     

Multiplicative effects on sentence comprehension for combined acoustic distortions.

This study was designed to explore the effect on speech comprehension of combining two types of signal distortion. A tape of clearly-enunciated sentences in quiet was distorted in each of four ways: low-pass (LP) filtering, time compression, interruption, and noise masking. Data are reported on a population of normal-hearing young men for multiple-choice answer tests of colloquial sentences of either LP filtered at 1, 2, 3, or 4 kHz, time compressed by computer at 250 words/min, interrupted (50 msec on--50 msec off), masked by speech-spectrum noise at +2 dB S/N, or given each of the 12 possible combinations of LP filtering plus the other three distortions. Individual distortion conditions were adjusted to reduce speech comprehension performance to about 90% accuracy, Low-pass filtering above 1 kHz reduced comprehension by no more than five to 10 percentage points, but when LP filtering was added to the other distortions in turn, latent effects were uncovered. The reduction in comprehension with the combined distortions was much greater than the simple additive effects of the distortion and LP filtering by themselves. For example, LP filtering above 2 kHz produced no measurable effect on sentence comprehension but this same distortion in combination with noise masking reduced performance from 89.4 to 59.7% correct (where 25% was chance). This study further validates the multiply-compounded nature of simultaneous types of distortion. The use of LP filtering extends the multiplicative principle to the simulated case of high-frequency hearing losses.     

瞬态诱发耳声发射频域分析的临床意义

目的：确定瞬态诱发耳声发射（ＴＥＯＡＥ）频域分析的应用价值。方法：对正常耳，噪声暴露耳，梅尼埃病耳，中耳病变耳及对侧抑制耳，进行ＴＥＯＡＥ频域测试。结果（１）噪声暴露耳，一些频率带的频信噪比降低，主频出现移动变化；（２）梅尼埃病一些甘油摄入后，频带信噪比增高，未引出ＴＥＯＡＥ的部分耳出现ＴＥＯＡＥ，主频也有移动变化；（３）中耳病变耳，不同的鼓室病变耳的频带信噪比降低，程度各异，其变化与声导纳改变有     

Use of adaptive digital signal processing to improve speech communication for normally hearing and hearing-impaired subjects

A two microphone adaptive digital noise cancellation technique was used to improve word-recognition ability of normally hearing and hearing-impaired subjects in the presence of varying amounts of multitalker speech babble noise and speech spectrum noise. Signal-to-noise ratios varied from -8 dB to +12 dB in 4 dB increments. The adaptive noise cancellation technique resulted in reducing both the speech babble and speech spectrum noises 18 to 22 dB. This reduction in noise resulted in average improvements in word recognition, at the poorest signal-to-noise ratios, ranging from 37% to 50% for the normally hearing subjects and 27% to 40% for the hearing-impaired subjects. Improvements in word recognition in the presence of speech babble noise as a result of adaptive filtering were just as large or larger than improvements found in the presence of speech spectrum noise. The amount of improvement of word-recognition scores was most pronounced at the least favorable signal-to-noise ratios.     

Improving performance in noise for hearing aids and cochlear implants using coherent modulation filtering

This study evaluated the maximal attainable performance of speech enhancement strategies based on coherent modulation filtering. An optimal adaptive coherent modulation filtering algorithm was designed to enhance known signals from a target talker in two-talker babble noise. The algorithm was evaluated in a closed-set, speech-recognition-in-noise task. The speech reception threshold (SRT) was measured using a one-down, one-up adaptive procedure. Five hearing-impaired subjects and five cochlear implant users were tested in three processing conditions: (1) original sounds; (2) fixed coherent modulation filtered sounds; and (3) optimal coherent modulation filtered sounds. Six normal-hearing subjects were tested with a 6-channel cochlear implant simulation of sounds processed in the same three conditions. Significant improvements in SRTs were observed when the signal was processed with the optimal coherent modulation filtering algorithm. There was no benefit when the signal was processed with the fixed modulation filter. The current study suggested that coherent modulation filtering might be a promising method for front-end processing in hearing aids and cochlear implants. An approach such as hidden Markov models could be used to generalize the optimal coherent modulation filtering algorithm to unknown utterances and to extend it to open-set speech.