空间插值在功率谱密度计算中的应用

 为了提高光学元件波前中频PSD计算的精度和有效频谱宽度,提出了填补波前无效数据的双线性插值法和抑制欠采样噪声的六采样点插值法。模拟计算和实验结果表明：双线性插值法有效地保证了填充数据与真实数据的一致性,抑制了零填充方法引入的虚假中高频信息,使得填补后的PSD与原始PSD较好地吻合;六采样点插值法有效地分离了信号和欠采样噪声,使得有效PSD频谱上限从1/2 Nyquist频率提高到Nyquist频率。     

基于可调谐激光的光谱辐射照度响应度定标

探测器的光谱辐射照(亮)度响应度是辐射定标中最重要的参数之一.传统的光谱辐射定标采用宽谱段光源和单色仪装置测量,新建的激光辐射测量装置采用激光和探测器测量,可以大大降低测量的不确定度.该装置首先将可调谐激光耦合进入积分球生成均匀的朗伯体单色光源,然后采用低温辐射计量传的标准陷阱探测器和面积已知的光阑,进行400～900...     

Representation of a Monotone Curve by a Contour with Regular Change in Curvature

The problem of modelling a smooth contour with a regular change in curvature representing a monotone curve with specified accuracy is solved in this article. The contour was formed within the area of the possible location of a convex curve, which can interpolate a point series. The assumption that if a sequence of points can be interpolated by a monotone curve, then the reference curve on which these points have been assigned is monotone, provides the opportunity to implement the proposed approach to estimate the interpolation error of a point series of arbitrary configuration. The proposed methods for forming a convex regular contour by arcs of ellipses and B-spline ensure the interpolation of any point series in parts that can be interpolated by a monotone curve. At the same time, the deflection of the contour from the boundaries of the area of the possible location of the monotone curve can be controlled. The possibilities of the developed methods are tested while solving problems of the interpolation of a point series belonging to monotone curves. The problems are solved in the CAD system of SolidWorks with the use of software application created based on the methods developed in the research work.     

结合幅度谱和功率谱字典的语音增强方法

从双路字典学习、噪声功率谱估计、语音幅度谱重构角度提出了一种改进的谱特征稀疏表示语音增强方法.在字典学习阶段,融合功率谱与幅度谱特征,采用区分性字典降低语音字典和噪声字典的相干性;在语音增强阶段,提出一种噪声功率谱估计方法对非平稳噪声进行跟踪估计;考虑到幅度谱和功率谱特征对不同噪声的适应程度不同,设计了语音重构权值表....     

定点脉冲线性预测编码方法研究*

在多脉冲线性预测编码的基础上,本文提出了位置无关脉冲搜索算法。该算法不需要搜索脉冲位置,而是根据给定的脉冲位置一次性解出脉冲幅度矢量。这就保证了得到的脉冲组合在最小二乘意义下是最优的,为改进合成语音质量提供了理论基础。进而在激励脉冲与位置无关的理论基础上,提出了定点脉冲线性预测编码方法。对所提出的算法在MATLAB下进行了仿真,仿真结果发现位置无关脉冲搜索算法得到的合成语音质量优于序贯法,编码时间也要比序贯法短。定点脉冲线性预测编码方法可以在2.7 kbps的编码速率下获得与G.729相近的合成语音。     

基于Voigt函数的拉曼成像插值方法

拉曼成像是拉曼光谱技术非常重要的一个环节,通过生成光谱数据的伪彩图像,可以得到采集区域中某物质组分的浓度和位置分布信息,当前,拉曼成像技术已经逐渐成为监测生物活性以及物质组分的最优解之一.为了得到清晰的成像效果,采集过程中的数据量不宜过小,否则成像效果差、锯齿感较重,从而导致视觉效果不好.但是,数据量的增加虽然可以得到...     

光纤激光气体氮化钛合金发射光谱及等离子体研究

激光气体氮化工艺可在钛合金表面快速生成氮化层,提高钛合金表面硬度和耐磨性,促进钛合金应用.采用光纤激光气体氮化Ti-6Al-4V合金,为了明确氮化过程光谱发射区是否形成等离子体,采用探针法检测了光谱发射区导电性;为了研究工艺参数对光谱特性、光谱发射区温度及等离子数量的影响,采用光谱仪采集了氮化过程发射光谱,并采用高速摄...     

A model for estimating ultrasound attenuation along the propagation path to the fetus from backscattered waveforms

Accurate estimates of the ultrasound pressure and/or intensity incident on the developing fetus on a patient-specific basis could improve the diagnostic potential of medical ultrasound by allowing the clinician to increase the transmit power while still avoiding the potential for harmful bioeffects. Neglecting nonlinear effects, the pressure/intensity can be estimated if an accurate estimate of the attenuation along the propagation path (i.e., total attenuation) can be obtained. Herein, a method for determining the total attenuation from the backscattered power spectrum from the developing fetus is proposed. The boundaries between amnion and either the fetus' skull or soft tissue are each modeled as planar impedance boundaries at an unknown orientation with respect to the sound beam. A mathematical analysis demonstrates that the normalized returned voltage spectrum from this model is independent of the planes orientation. Hence, the total attenuation can be estimated by comparing the location of the spectral peak in the reflection from the fetus to the location of the spectral peak in a reflection obtained from a rigid plane in a water bath. The independence of the attenuation estimate and plane orientation is then demonstrated experimentally using a Plexiglas plate, a rat's skull, and a tissue-mimicking phantom.     

汉语耳语标准频谱的测量与计算

提出了与GB7348-87《耳语标准频谱》不同的汉语耳语功率谱密度级随频率的变化关系。在消声室中测量以提高测量信噪比,使用实时分析仪测量单个人耳语发音的长期声压频谱,并且对每个人的长期声压频谱做自归一化,通过数学方法将多个样本"混录",计算出汉语耳语的功率谱密度级。汉语耳语标准频谱的测量和计算结果可为一切产生、传输、接收和处理汉语耳语信号的系统及电声器件的设计提供依据。      

Perceptual linear predictive (PLP) analysis of speech

A new technique for the analysis of speech, the perceptual linear predictive (PLP) technique, is presented and examined. This technique uses three concepts from the psychophysics of hearing to derive an estimate of the auditory spectrum: (1) the critical-band spectral resolution, (2) the equal-loudness curve, and (3) the intensity-loudness power law. The auditory spectrum is then approximated by an autoregressive all-pole model. A 5th-order all-pole model is effective in suppressing speaker-dependent details of the auditory spectrum. In comparison with conventional linear predictive (LP) analysis, PLP analysis is more consistent with human hearing. The effective second formant F2' and the 3.5-Bark spectral-peak integration theories of vowel perception are well accounted for. PLP analysis is computationally efficient and yields a low-dimensional representation of speech. These properties are found to be useful in speaker-independent automatic-speech recognition.     

Speech dereverberation method based on spectral subtraction and spectral line enhancement

Speech signals recorded with a distant microphone usually are interfered by the spatial reverberation in the room, which severely degrades the clarity and intelligibility of speech. A speech dereverberation method based on spectral subtraction and spectral line enhancement is proposed in this paper. Following the generalized statistical reverberation model, the power spectrum of late reverberation is estimated and removed from the reverberation speech by the spectral subtraction method. Then, according to the human auditory model, a spectral line enhancement technique based on adaptive post-filtering is adopted to further eliminate the reverberant components between adjacent speech formants. The proposed method can effectively suppress the spatial reverberation and improve the auditory perception of speech. The subjective and objective evaluation results reveal that the perceptual quality of speech is greatly improved by the proposed method.     

Sinusoidal modeling for nonstationary voiced speech based on a local vector transform

A voiced speech signal can be expressed as a sum of sinusoidal components of which instantaneous frequency and amplitude continuously vary with time. Determining these parameters from the input, the time-varying characteristics are crucial error sources for the algorithms, which assume their stationarity within a local analysis segment. To overcome this problem, a new method is proposed, local vector transform (LVT), which can determine instantaneous frequency and amplitude for nonstationary sinusoids. The method does not assume the local stationarity. The effectiveness of LVT was examined in parameter determination for synthesized and naturally uttered speech signals. The instantaneous frequency for the first harmonic component was determined with an accuracy almost equal to that of the time-corrected instantaneous frequency method and higher accuracy than that of spectral peak-picking, autocorrelation, and cepstrum. The instantaneous amplitude was also determined accurately by LVT while considerable errors were left in the other algorithms. The signal reconstructed from the determined parameters by LVT agreed well with the corresponding component of voiced speech. These results suggest that the method is effective for analyzing time-varying voiced speech signals.     

Parameter interpolation in speech synthesis

A comparison has been made of the transition properties of six types of speech synthesizer parameters: serial resonance, prediction coefficients, reflection coefficients, area functions, parallel resonance, and, finally, a simple set of articulatory parameters. The first four synthesizers are formally equivalent and can be made to produce identical steady-state sounds (targets). The last two involve approximations, but achieve similar targets. Formant paths between targets will differ according to the parameter type used during interpolation. Each type was tested on nonsense words spanning a wide range of parameter values. Linear interpolation of synthesizer parameters was used to determine a path between target values. The resultant data were then converted to formant values and plotted as a spectrographic (frequency versus time) representation. Small differences in formant frequency (versus linear transitions of formant frequency and bandwidth) were common, and some quite large differences in formant bandwidths were observed in certain cases.     

基于光谱线性拟合的宽波段光源波动补偿方法

宽波段光源在工作过程中,光强通常会随供电电源输出功率的变化而波动。宽波段光源波段内各个波长的光谱强度将会发生不同程度的波动。为解决光源光强波动时其波段内各个波长光谱强度的补偿问题,提出了一种基于光谱线性拟合的补偿方法。使用这种方法,只需测量光源波段光强的变化,就可以补偿各个波长光谱强度的波动。通过分析理想黑体全波段辐射出射度与光谱辐射出射度的近似线性关系。建立了宽波段光源波段光强与光谱强度的线性模型。实验装置主要由卤素灯珠、光源电源、光阑、光谱仪及计算机构成。调节电源的输出功率,得到一组卤素灯珠在不同输入功率下的相对光谱强度。测量不同功率下卤素灯珠光谱强度来验证该方法补偿效果。按线性关系拟合了卤素灯珠光谱强度与其波段光强关系式,并分析了拟合误差。实验表明：卤素灯珠的光谱强度与其波段光强具有很好的线性关系,可以用波段光强按线性关系来补偿其光谱强度的波动。随着卤素灯珠输入功率的增大,补偿后的光谱强度的相对误差绝对值下降。在卤素灯珠输入功率范围内,使用该方法补偿的光谱强度在绝大部分(92%)波长下相对误差绝对值可在5%以内。     

基于相位相关的匀速直线运动模糊图像位移参数估计

为估算匀速直线运动模糊图像的运动参数,提出了一种基于相位相关分析的图像配准方法。该方法利用傅里叶变换的平移特性,对产生平移的目标图像进行傅里叶变换并计算归一化互功率谱,其傅里叶逆变换对应二维脉冲函数,通过计算脉冲函数峰值坐标获取位移图像之间的亚像元级位移量。结合相位相关配准原理和线性空间不变退化模型,给出了匀速直线运动点扩散函数及其光学传递函数的数学描述;讨论了匀速直线运动模糊对相位相关配准结果的影响,证明了图像经过匀速直线运动退化后,位移图像之间归一化互功率谱具有不变性。实验结果表明：动态运动模糊图像最大检测误差为0.489 pixel,标准差为0.16 pixel。     

Improving the performance of speech waveform synthesis using WaveNet fused with phase information

Compared with phase spectrum, magnitude spectrum can represent most speech information, hence many speech processing tasks pay much attention on manipulating magnitude spectrum and use the imperfect vocoder parameters or mismatched phase spectrum to synthesize the waveform, which leads to an obvious distortion of speech quality. To address this problem, a modified version of Wave Net model fused with phase information is proposed to synthesize the speech with higher quality. In the Wave Net model, the original or processed phase spectrum of speech and the enhanced magnitude spectrum are concatenated as the condition input, and then the predicted speech waveform is generated directly from this input, which is a kind of fusion feature. The proposed method can realize the effective utilization of the phase information and is verified in two tasks including voice conversion(VC) and bone-conducted speech enhancement(BSE). Two kinds of phase spectrum, the modified group delay(MGD)spectrum and the instantaneous frequency deviation spectrum, are compared comprehensively in the simulation experiments, and the influence of the fusion feature on the bandwidth extension Wave Net model and the teacher-student Wave Net model is also explored. In VC experiments, the A/B test shows the generated speech using the teacher-student Wave Net model is much better than using the STRAIGHT vocoder. In BSE experiments, the results show that,using the bandwidth extension Wave Net model via the feature fused with MGD spectrum, the mean opinion score(MOS) of the enhanced speech increases by 54.3% compared with the original bone-conducted speech. All the results demonstrate that the phase-fused condition input can supplement single magnitude spectrum efficiently and help the Wave Net vocoder achieve promising improvement on the quality of the synthesized speech.     

基于数字信号处理的嗓音控制开关(VOX)算法研究

本文提出了一种基于数字谱分析的嗓音控制开关(VOX，Voice—Operated Transmit)的新算法，该算法简单、实用，在某种程度上克服了传统VOX算法的结构复杂、参数难调等局限，对噪声的鲁棒性也较好，而且易于用数字信号处理实现。首先利用信号功率谱二次处理，提取出语音的平均幅度包络，然后对所得包络进行阈值处理、限幅放大，最后就得到VOX函数。理论分析和计算机模拟结果表明，该算法不仅能较为准确地提取出语音波形的平均幅度包络，而且能工作在较低的信噪比条件下。     

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.     

The synergy between speech production and perception

Speech intelligibility is known to be relatively unaffected by certain deformations of the acoustic spectrum. These include translations, stretching or contracting dilations, and shearing of the spectrum (represented along the logarithmic frequency axis). It is argued here that such robustness reflects a synergy between vocal production and auditory perception. Thus, on the one hand, it is shown that these spectral distortions are produced by common and unavoidable variations among different speakers pertaining to the length, cross-sectional profile, and losses of their vocal tracts. On the other hand, it is argued that these spectral changes leave the auditory cortical representation of the spectrum largely unchanged except for translations along one of its representational axes. These assertions are supported by analyses of production and perception models. On the production side, a simplified sinusoidal model of the vocal tract is developed which analytically relates a few "articulatory" parameters, such as the extent and location of the vocal tract constriction, to the spectral peaks of the acoustic spectra synthesized from it. The model is evaluated by comparing the identification of synthesized sustained vowels to labeled natural vowels extracted from the TIMIT corpus. On the perception side a "multiscale" model of sound processing is utilized to elucidate the effects of the deformations on the representation of the acoustic spectrum in the primary auditory cortex. Finally, the implications of these results for the perception of generally identifiable classes of sound sources beyond the specific case of speech and the vocal tract are discussed.     

Spectral distortion of dual-comb spectrometry due to repetition rate fluctuation

Dual-comb spectrometry suffers the fluctuations of parameters in combs. We demonstrate that the repetition rate is more important than any other parameter, since the fluctuation of the repetition rate leads to a change of difference in the repetition rate between both combs, consequently causing the conversion factor variation and spectral frequency misalignment. The measured frequency noise power spectral density of the repetition rate exhibits an integrated residual frequency modulation of 1.4 Hz from 1 Hz to 100 k Hz in our system. This value corresponds to the absorption peak fluctuation within a root mean square value of 0.19 cm~(-1) that is verified by both simulation and experimental result.Further, we can also simulate spectrum degradation as the fluctuation varies. After modifying misaligned spectra and averaging, the measured result agrees well with the simulated spectrum based on the GEISA database.