语后聋人工耳蜗使用者电刺激听觉部位音调感知研究

系统地研究了人工耳蜗植入者的电刺激听觉部位音调感知,全面地探讨了部位音调感知与人工耳蜗植入者言语识别和音乐感知的关系。4位成人语后聋人工耳蜗植入者参与了该研究。通过电极音调排序测试度量植入者的部位音调感知能力。言语能力测试和音乐音高分辨测试分别用米考察植入者的言语识别和音乐感知能力。结果显示,随着电极刺激部位从蜗尖移向蜗底,所有受试者均可获得从"低"到"高"的音调感知变化,但个体差异较大。受试者的言语识别结果与其电刺激听觉部位音调感知能力相关,但受到天花板效应影响,对应关系并不明显。受试者的音乐音高分辨成绩与其电刺激听觉部位音调感知能力呈较好的对应关系。结果表明,当前人工耳蜗声音编码策略所传递的声信号特征已可使植入者获得良好的言语识别效果;且安静环境下言语识别对植入者的部位音调感知能力要求不高。但当前的声音编码策略并未能有效对音乐信号进行编码;植入者在理解音乐这类复杂声信号时,其电刺激听觉部位音调感知能力一定程度决定了其听音效果。      

基于相位/强度调制的伪毫米波超宽带信号的光学辅助探测（英文）

针对伪毫米波超宽带信号脉冲宽度窄且信号强度低而不易探测的特点,本文提出了两种光学辅助包络探测方案.将伪毫米波超宽带信号调制到光载波上,利用光纤布喇格光栅带通滤波器滤出调制器输出的第一边带信号,通过光电探测器和低通滤波器便可以得到用于判决的伪毫米波超宽带信号的包络信息.数学推导和仿真分析了分别采用相位调制器和强度调制器对输出包络信号振幅的影响.研究结果表明:强度调制器需要偏置控制电路来稳定偏置点的位置,当强度调制器工作在最小传输点时,得到的包络信号幅值最大;相位调制器无需偏置控制电路,其输出信号包络的幅值等于使用强度调制器的系统可得到的最大值;使用相位调制器的系统较使用强制调制器的系统具有结构更简单和插入损耗更低的特点.     

语音中相位的听觉感知实验研究

人的听觉对语音信号中相位的感知比较迟钝,因而对语音信号进行处理和编码时常常不关心相位失真。实际上,相位失真到一定程度时会明显导致语音质量的下降。为了取得高质量的声码器,语音谱分量的相位信息是不能不考虑的。本文通过主观听觉测试实验研究了语音信号的短时Fourier变换相位谱对人的听觉感知的影响。测试结果表明: (1)如果完全舍弃原相位信息,则得到的重建语音含有很强的噪声且自然度很差; (2)不论舍弃高频段还是低频段的相位信息,均能导致听觉感知差异; (3)当相位的量阶小于π/7时,人的听觉系统将分辨不出重建语音和原始语音之间存在的差异.     

低强度周期量级脉冲驱动排列分子的非次序双电离

本文利用三维经典系综模型研究了低强度周期量级脉冲驱动排列分子的非次序双电离. 结果表明, 电子对的关联特性强烈地依赖于分子的排列方向和激光脉冲的载波包络相位; 垂直分子反关联电子对的比例总是高于平行分子反关联电子对的比例; 当载波包络相位从0到π 逐渐增加时, 反关联电子对的数目先增加再减少; 对于平行分子, 电子对的释放总是以正关联为主; 而垂直分子的主导关联模式则依赖于激光脉冲的载波包络相位, 当载波包络相位为0.3π-0.7π之间时, 电子对以反关联释放为主, 其他相位下以正关联为主. 本文利用分子势能曲线和电子返回能量很好地解释了电子关联特性对分子排列方向和载波包络相位的依赖关系.     

飞秒钛宝石放大激光脉冲的载波包络相位测量与控制

通过对重复频率为1 kHz的放大飞秒激光脉冲的光谱干涉实验,结合傅里叶变换进行了载波包络相位漂移的实验研究.在此基础上利用锁相环反馈控制技术实现了对载波包络相位的精密锁定,锁定后的激光脉冲稳态相位均方根误差小于80mrad,锁定时间超过3h.同时在理论上分析了光谱干涉测量放大激光脉冲载波包络相位的原理,给出了光谱干涉信号与载波包络相位的关系. 关键词： 飞秒钛宝石放大器 载波包络相位 光谱干涉 超连续     

基于离散余弦变换的语音压缩采样和编码算法

针对语音无线通信中带宽资源受限的问题,提出基于压缩采样的低速率语音编码算法。以基尼系数为指标,比较不同稀疏变换域下语音信号的稀疏性,分析常见重构算法对语音信号压缩采样观测信号的重构特性。对标准耳蜗滤波器——伽马啁啾滤波器组的参数进行研究,并以梯度投影稀疏重建(GPSR)算法重构语音信号。利用语音质量感知评估(PESQ)、信噪比和主观听觉测试,对编解码后的合成语音信号进行了质量评估。实验表明,基于压缩感知的语音编码器以4 kbps的低速率对语音进行编码时,PESQ得分可达到3.16,计算复杂度相对较低,可以用于实际的语音编码环境。     

兴奋模式下谐波复合音音高感知机制的研究

通过测量谐波复合音的基频辨别阈,探讨中等"高次谐波"的音高感知是否依赖于谐波的可分离性,以及掩蔽音对实验结果的影响。实验方法:在目标音单独存在或目标音与掩蔽音混合时,将刺激通过高、中、低三个带通滤波器以获得不同的谐波可分离度。实验刺激设计为5种基频差异和4种相位组合。五名被试均为年轻人,纯音听阈≤15 dB HL。研究结果发现:谐波复合音的基频辨别阈随着信号频段的上移而增大;目标音和掩蔽音的基频差异对基频辨别阈有显著影响;但相位影响不显著。结论:谐波的可分离性对基频辨别阈有显著影响,但中等"高次谐波"的音高感知不依赖于可分离性;混合音的大部分音高感知结果与兴奋模式的峰值大小密切相关。      

一种自适应变阶谱减降噪算法在电子耳蜗中的应用

由于传统谱减语音增强存在残留的"音乐噪声",因此基于传统谱减法降噪的电子耳蜗(CI)感知的声音品质也会受到影响.为提高CI的抗噪性,本文提出了一种自适应变阶谱减算法,并将该方法应用于电子耳蜗的语音增强中.根据CI电极对应的频带关系,该算法先对采集的带噪声音信号功率谱进行Bark子带划分,并在每个Bark子带中根据信噪比的变化进行谱减阶数和系数的自适应调节,使各子带噪声更均衡地去除,基本消除了传统方法存在的"音乐噪声".基于该算法的电子耳蜗ACE仿真实验及测听结果表明,与传统谱减法相比,改进的算法能更好地抑制背景噪声和残留噪声,仿真得到的CI合成音感知更好和更清晰.     

共振条件下载波包络相位效应对阈上电离谱的影响

本文通过求解三维含时薛定谔方程, 从理论上研究了共振条件下氢原子的光电子能谱与角分布随着激光脉冲载波包络相位(CEP)的变化规律. 研究结果表明: 在共振位置附近, 光电子能谱的强度与CEP有强烈的依赖关系; 进一步对各个分波的分析, 可以确定共振电离初态的角量子数. 此外, 利用光电子的角分布信息, 可以对多周期激光脉冲的初始相位信息进行高精度探测. 关键词： 阈上电离 超短脉冲 载波包络相位     

任意复包络信号的匀速运动目标回波脉间补偿及相干积累

现代声呐、水下制导等水声探测系统常常使用窄带脉冲、调制、编码、伪随机等种类繁多的发射信号波形来满足低信噪比检测、高分辨估计、抗干扰、主动隐蔽探测的应用需求.针对这一情况,本文研究了任意信号的长时间积累问题,给出了一种任意复包络信号的匀速运动目标回波脉间补偿及相干积累检测方法.通过构建任意发射信号波形的广义模糊函数,将匹配滤波器输出表示为所构造的广义模糊函数,使得任意复包络信号的脉压波形不仅能够用统一的数学模型来表述和计算,而且能够提供多脉冲回波的距离走动信息和多普勒频移信息,为多脉冲距离位置对齐和Fourier变换(FFT)积累提供了依据.对于用广义模糊函数表示的匹配滤波器输出,采用Keystone变换将复包络对齐,消除了距离走动,采用FFT补偿多普勒频移项,实现了任意复包络信号的长时间相干积累.对于水下探测中使用的连续波信号、线性调频信号以及复杂的m序列编码信号、Costas跳频编码信号波形进行了信号积累及检测的计算机仿真,验证了任意复包络信号的匀速运动目标回波脉间补偿及相干积累的正确性.消声水池实验验证了该方法的有效性.     

Effects of interaural time differences in fine structure and envelope on lateral discrimination in electric hearing

Bilateral cochlear implant (CI) listeners currently use stimulation strategies which encode interaural time differences (ITD) in the temporal envelope but which do not transmit ITD in the fine structure, due to the constant phase in the electric pulse train. To determine the utility of encoding ITD in the fine structure, ITD-based lateralization was investigated with four CI listeners and four normal hearing (NH) subjects listening to a simulation of electric stimulation. Lateralization discrimination was tested at different pulse rates for various combinations of independently controlled fine structure ITD and envelope ITD. Results for electric hearing show that the fine structure ITD had the strongest impact on lateralization at lower pulse rates, with significant effects for pulse rates up to 800 pulses per second. At higher pulse rates, lateralization discrimination depended solely on the envelope ITD. The data suggest that bilateral CI listeners benefit from transmitting fine structure ITD at lower pulse rates. However, there were strong interindividual differences: the better performing CI listeners performed comparably to the NH listeners.     

Recognition of spectrally asynchronous speech by normal-hearing listeners and Nucleus-22 cochlear implant users

This experiment examined the effects of spectral resolution and fine spectral structure on recognition of spectrally asynchronous sentences by normal-hearing and cochlear implant listeners. Sentence recognition was measured in six normal-hearing subjects listening to either full-spectrum or noise-band processors and five Nucleus-22 cochlear implant listeners fitted with 4-channel continuous interleaved sampling (CIS) processors. For the full-spectrum processor, the speech signals were divided into either 4 or 16 channels. For the noise-band processor, after band-pass filtering into 4 or 16 channels, the envelope of each channel was extracted and used to modulate noise of the same bandwidth as the analysis band, thus eliminating the fine spectral structure available in the full-spectrum processor. For the 4-channel CIS processor, the amplitude envelopes extracted from four bands were transformed to electric currents by a power function and the resulting electric currents were used to modulate pulse trains delivered to four electrode pairs. For all processors, the output of each channel was time-shifted relative to other channels, varying the channel delay across channels from 0 to 240 ms (in 40-ms steps). Within each delay condition, all channels were desynchronized such that the cross-channel delays between adjacent channels were maximized, thereby avoiding local pockets of channel synchrony. Results show no significant difference between the 4- and 16-channel full-spectrum speech processor for normal-hearing listeners. Recognition scores dropped significantly only when the maximum delay reached 200 ms for the 4-channel processor and 240 ms for the 16-channel processor. When fine spectral structures were removed in the noise-band processor, sentence recognition dropped significantly when the maximum delay was 160 ms for the 16-channel noise-band processor and 40 ms for the 4-channel noise-band processor. There was no significant difference between implant listeners using the 4-channel CIS processor and normal-hearing listeners using the 4-channel noise-band processor. The results imply that when fine spectral structures are not available, as in the implant listener's case, increased spectral resolution is important for overcoming cross-channel asynchrony in speech signals.     

Effects of cochlear implant processing and fundamental frequency on the intelligibility of competing sentences

Speech perception in the presence of another competing voice is one of the most challenging tasks for cochlear implant users. Several studies have shown that (1) the fundamental frequency (F0) is a useful cue for segregating competing speech sounds and (2) the F0 is better represented by the temporal fine structure than by the temporal envelope. However, current cochlear implant speech processing algorithms emphasize temporal envelope information and discard the temporal fine structure. In this study, speech recognition was measured as a function of the F0 separation of the target and competing sentence in normal-hearing and cochlear implant listeners. For the normal-hearing listeners, the combined sentences were processed through either a standard implant simulation or a new algorithm which additionally extracts a slowed-down version of the temporal fine structure (called Frequency-Amplitude-Modulation-Encoding). The results showed no benefit of increasing F0 separation for the cochlear implant or simulation groups. In contrast, the new algorithm resulted in gradual improvements with increasing F0 separation, similar to that found with unprocessed sentences. These results emphasize the importance of temporal fine structure for speech perception and demonstrate a potential remedy for difficulty in the perceptual segregation of competing speech sounds.     

The effects of the addition of low-level, low-noise noise on the intelligibility of sentences processed to remove temporal envelope information

The intelligibility of sentences processed to remove temporal envelope information, as far as possible, was assessed. Sentences were filtered into N analysis channels, and each channel signal was divided by its Hilbert envelope to remove envelope information but leave temporal fine structure (TFS) intact. Channel signals were combined to give TFS speech. The effect of adding low-level low-noise noise (LNN) to each channel signal before processing was assessed. The addition of LNN reduced the amplification of low-level signal portions that contained large excursions in instantaneous frequency, and improved the intelligibility of simple TFS speech sentences, but not more complex sentences. It also reduced the time needed to reach a stable level of performance. The recovery of envelope cues by peripheral auditory filtering was investigated by measuring the intelligibility of 'recovered-envelope speech', formed by filtering TFS speech with an array of simulated auditory filters, and using the envelopes at the output of these filters to modulate sinusoids with frequencies equal to the filter center frequencies (i.e., tone vocoding). The intelligibility of TFS speech and recovered-envelope speech fell as N increased, although TFS speech was still highly intelligible for values of N for which the intelligibility of recovered-envelope speech was low.     

Coding of the fundamental frequency in continuous interleaved sampling processors for cochlear implants

In this study the perception of the fundamental frequency (F0) of periodic stimuli by cochlear implant users is investigated. A widely used speech processor is the Continuous Interleaved Sampling (CIS) processor, for which the fundamental frequency appears as temporal fluctuations in the envelopes at the output. Three experiments with four users of the LAURA (Registered trade mark of Philips Hearing Implants, now Cochlear Technology Centre Europe) cochlear implant were carried out to examine the influence of the modulation depth of these envelope fluctuations on pitch discrimination. In the first experiment, the subjects were asked to discriminate between two SAM (sinusoidally amplitude modulated) pulse trains on a single electrode channel differing in modulation frequency ( deltaf = 20%). As expected, the results showed a decrease in the performance for smaller modulation depths. Optimal performance was reached for modulation depths between 20% and 99%, depending on subject, electrode channel, and modulation frequency. In the second experiment, the smallest noticeable difference in F0 of synthetic vowels was measured for three algorithms that differed in the obtained modulation depth at the output: the default CIS strategy, the CIS strategy in which the F0 fluctuations in the envelope were removed (FLAT CIS), and a third CIS strategy, which was especially designed to control and increase the depth of these fluctuations (F0 CIS). In general, performance was poorest for the FLAT CIS strategy, where changes in F0 are only apparent as changes of the average amplitude in the channel outputs. This emphasizes the importance of temporal coding of F0 in the speech envelope for pitch perception. No significantly better results were obtained for the F0 CIS strategy compared to the default CIS strategy, although the latter results in envelope modulation depths at which sub-optimal scores were obtained in some cases of the first experiment. This indicates that less modulation is needed if all channels are stimulated with synchronous F0 fluctuations. This hypothesis is confirmed in a third experiment where subjects performed significantly better in a pitch discrimination task with SAM pulse trains, if three channels were stimulated concurrently, as opposed to only one.     

Enhancing Chinese tone recognition by manipulating amplitude envelope: implications for cochlear implants

Tone recognition is important for speech understanding in tonal languages such as Mandarin Chinese. Cochlear implant patients are able to perceive some tonal information by using temporal cues such as periodicity-related amplitude fluctuations and similarities between the fundamental frequency (F0) contour and the amplitude envelope. The present study investigates whether modifying the amplitude envelope to better resemble the F0 contour can further improve tone recognition in multichannel cochlear implants. Chinese tone and vowel recognition were measured for six native Chinese normal-hearing subjects listening to a simulation of a four-channel cochlear implant speech processor with and without amplitude envelope enhancement. Two algorithms were proposed to modify the amplitude envelope to more closely resemble the F0 contour. In the first algorithm, the amplitude envelope as well as the modulation depth of periodicity fluctuations was adjusted for each spectral channel. In the second algorithm, the overall amplitude envelope was adjusted before multichannel speech processing, thus reducing any local distortions to the speech spectral envelope. The results showed that both algorithms significantly improved Chinese tone recognition. By adjusting the overall amplitude envelope to match the F0 contour before multichannel processing, vowel recognition was better preserved and less speech-processing computation was required. The results suggest that modifying the amplitude envelope to more closely resemble the F0 contour may be a useful approach toward improving Chinese-speaking cochlear implant patients' tone recognition.     

Speech processing studies using an acoustic model of a multiple-channel cochlear implant

The speech perception of two multiple-channel cochlear implant patients was compared with that of three normally hearing listeners using an acoustic model of the implant for 22 different speech tests. The tests used included a minimal auditory capabilities battery, both closed-set and open-set word and sentence tests, speech tracking and a 12-consonant confusion study using nonsense syllables. The acoustic model represented electrical current pulses by bursts of noise and the effects of different electrodes were represented by using bandpass filters with different center frequencies. All subjects used a speech processor that coded the fundamental voicing frequency of speech as a pulse rate and the second formant frequency of speech as the electrode position in the cochlea, or the center frequency of the bandpass filter. Very good agreement was found for the two groups of subjects, indicating that the acoustic model is a useful tool for the development and evaluation of alternative cochlear implant speech processing strategies.     

Features of stimulation affecting tonal-speech perception: implications for cochlear prostheses

Tone languages differ from English in that the pitch pattern of a single-syllable word conveys lexical meaning. In the present study, dependence of tonal-speech perception on features of the stimulation was examined using an acoustic simulation of a CIS-type speech-processing strategy for cochlear prostheses. Contributions of spectral features of the speech signals were assessed by varying the number of filter bands, while contributions of temporal envelope features were assessed by varying the low-pass cutoff frequency used for extracting the amplitude envelopes. Ten normal-hearing native Mandarin Chinese speakers were tested. When the low-pass cutoff frequency was fixed at 512 Hz, consonant, vowel, and sentence recognition improved as a function of the number of channels and reached plateau at 4 to 6 channels. Subjective judgments of sound quality continued to improve as the number of channels increased to 12, the highest number tested. Tone recognition, i.e., recognition of the four Mandarin tone patterns, depended on both the number of channels and the low-pass cutoff frequency. The trade-off between the temporal and spectral cues for tone recognition indicates that temporal cues can compensate for diminished spectral cues for tone recognition and vice versa. An additional tone recognition experiment using syllables of equal duration showed a marked decrease in performance, indicating that duration cues contribute to tone recognition. A third experiment showed that recognition of processed FM patterns that mimic Mandarin tone patterns was poor when temporal envelope and duration cues were removed.     

The effect of interaural differences in envelope shape on the perceived location of sounds (L)

Users of bilateral cochlear implants and a cochlear implant combined with a contralateral hearing aid are sensitive to interaural time differences (ITDs). The way cochlear implant speech processors work and differences between modalities may result in interaural differences in shape of the temporal envelope presented to the binaural system. The effect of interaural differences in envelope shape on ITD sensitivity was investigated with normal-hearing listeners using a 4?kHz pure tone modulated with a periodic envelope with a trapezoid shape in each cycle. In one ear the onset segment of the trapezoid was transformed by a power function. No effect on the just noticeable difference in ITD was found with an interaural difference in envelope shape, but the ITD for a centered percept was significantly different across envelope shape conditions.     

The ability of listeners to use recovered envelope cues from speech fine structure

Recent work has demonstrated that auditory filters recover temporal-envelope cues from speech fine structure when the former were removed by filtering or distortion. This study extended this work by assessing the contribution of recovered envelope cues to consonant perception as a function of the analysis bandwidth, when vowel-consonant-vowel (VCV) stimuli were processed in order to keep their fine structure only. The envelopes of these stimuli were extracted at the output of a bank of auditory filters and applied to pure tones whose frequency corresponded to the original filters' center frequencies. The resulting stimuli were found to be intelligible when the envelope was extracted from a single, wide analysis band. However, intelligibility decreases from one to eight bands with no further decrease beyond this value, indicating that the recovered envelope cues did not play a major role in consonant perception when the analysis bandwidth was narrower than four times the bandwidth of a normal auditory filter (i.e., number of analysis bands > or =8 for frequencies spanning 80 to 8020 Hz).