Frequency response testing in a noisy environment or with a limited power supply

The current trend in dynamic testing is towards short duration, transient testing in the field followed by detailed computer analysis of the data at some later time. For many applications, tests must be carried out in a noisy environment, such as in frequency response studies of operating machinery and vibration tests on buildings or ships where the excitation power source is small and produces a low level response compared with ambient vibration levels. The use of time domain averaging is demonstrated and it is shown to be a suitable signal enhancement procedure for testing in poor signal to noise ratio conditions, when using a transient excitation technique. The limitations and accuracy of the technique are discussed and examples of practical application are given.     

铷原子气体自旋噪声谱的测量与改进

利用自主设计并制作的基于现场可编程门阵列的实时傅里叶变换采集卡(FFTsDAC),采用线偏振光检测碱金属铷原子气样品中的自旋随机涨落(即自旋噪声谱).详细讨论了背景噪声以及自旋噪声随探测光光强的变化关系,证实了自旋噪声来自于系统中自旋的随机涨落.对比了两种FFTsDAC(8 bit采样的FFTsDACl和12 bit采样的FFTsDAC2)的测量性能,分析了影响实验信噪比的因素.FFTsDAC2具有更高的测量效率和采样深度以及更长的单次采样时间,因而具有更高的信噪比和更好的频率分辨率,与数值模拟的结果一致.     

FTIR被动遥测信噪比优化及在变电站SF6泄漏检测应用

被动FTIR红外遥测技术可应用于气体泄漏检测,气体检测下限与仪器信噪比密切相关。仪器信噪比与测量参数有关,如光谱分辨率、采样频率、积分时间、平均次数等,如何结合实际应用优化参数组合以实现最佳信噪比,目前还缺少系统分析。针对这一问题,从理论上分析了这些参数与信噪比关系,并归类为三个方面：（1）光谱分辨率,引用了Roland Harig给出的结论,当光谱分辨率低于特征峰半高全宽时,信噪比不变,但为了避免背景气体的交叉干扰,分辨率不宜设置过低,需要结合实际应用综合考虑;（2）降低采样频率能够减少计算量,缩小光谱范围,但采样频率和光谱范围与信噪比无关;（3）在积分时间和光谱平均次数方面,同样时间条件下,多次干涉图采集会引入零光程差的采样误差,使得噪声大于理论计算值,因而长积分时间获得信噪比优于多次平均。开展了六氟化硫（SF₆）泄漏检测实验,根据SF₆特征峰宽度选择4 cm-1分辨率,20 kHz采样频率兼顾了信噪比和检测时间,并利用FTIR巡检系统发现变电站泄漏点,证明了方法的有效性。     

The feasibility of maximum length sequences to reduce acquisition time of the middle latency response

Maximum length sequences (MLS) have been used to improve the signal-to-noise ratio (SNR) of otoacoustic emissions [Thornton, J. Acoust. Soc. Am. 94, 132-136 (1993)] and the auditory brainstem response [Thornton and Slaven, Br. J. Audiol. 27, 205-210 (1993)]. By implication, a shorter recording time would be required to give equal signal-to-noise ratio (SNR). This study aimed to establish whether it is also possible to improve the SNR of the auditory-evoked potential termed the middle latency response (MLR) using maximum length sequences (MLS). Recordings of 180 s each were made using a conventional recording rate and MLS rates of 42, 89, and 185 clicks/s. Three different stimulus intensities were used in the range 30 to 70 dB nHL. The rate of 89 clicks/s was found to produce most improvement in SNR for both the Na-Pa region of the MLR and the Na-Pb region. This improvement in SNR using MLS implies that an MLS rate of 89 clicks/s would produce a fourfold reduction in recording time for equal SNR over conventional recording for the Pa-Nb region of the MLR at a stimulus intensity of 70 dB nHL. The latency of the Nb wave was found to reduce significantly using MLS. An MLR could not be recorded from every subject in this study, but more subjects had an identifiable response for MLS than for conventional recordings. Use of MLS to record the MLR appears to offer the potential for reduction in test time and better wave identification.     

单载波频域均衡中的水声信道频域响应与噪声估计

本文对单载波频域均衡(SC-FDE)中的信道和噪声估计进行研究,理论分析了形成误码率平台的可能原因,提出了基于Chu序列的联合信道时频域响应和噪声功率估计算法.利用水声信道响应的稀疏特性和门限确定信道能量集中区域,进行信道估计的去噪处理和噪声功率计算.借助于水声射线模型对所提出的算法进行了仿真,分析了噪声估计对SC-FDE性能的重要影响.仿真结果表明,所提出的信道和噪声估计算法可以有效减缓或消除SC-FDE中的误码率平台.     

Signal-to-noise ratio and aberration statistics in ocular aberrometry

We define a signal-to-noise ratio (SNR) for eye aberrometry in terms of the sensor geometry, measurement noise, and population statistics. The overall estimation error is composed of three main contributions: the bias in the estimated modes, the truncation error, and the error due to the noise propagation. This last term can be easily parametrized by the proposed SNR. We compute the overall error as well as the magnitude of its three components for a typical sensor configuration, population statistics, and different SNR. We show that there are an optimum number of Zernike aberration modes to be retrieved in each case.     

一种应用于心磁噪声抑制的选择性平均方法研究

在心磁信号探测中, 抑制环境噪声是提取心磁信号的关键. 为了提高心磁信号的信噪比, 信号平均得到了广泛地使用. 然而, 由于局部干扰噪声的存在, 对整段数据进行平均的传统方法不可避免地会带来心磁信号的失真. 本文通过采取模版匹配的方式, 提出了一种选择性平均方法. 结果显示, 本方法能有效剔除带有低频波动、脉冲毛刺等干扰的数据段, 获取高质量的平均心磁信号. 关键词： 心磁 信号平均 噪声抑制 信噪比     

Effect of listening level and background noise on the subjective decay rate of room impulse responses: Using time-varying loudness to model reverberance

Room impulse responses (RIRs) are used very widely to characterize the acoustic conditions of rooms, such as in the derivation of reverberation time, early decay time and clarity index. This study investigates the subjective decay rate (or reverberance) of RIRs when directly listened to (rather than convolved with a dry signal such as speech or music). Through a subjective experiment, it investigates the effects of gain (or listening level) and background noise level on the reverberance of RIRs that had been measured in three concert auditoria. The task of the experiment was to match the decay rate of RIRs to that of a reference RIR by ear, by adjusting the RIRs’ exponential decay rate. Based on objective loudness modeling, gain should have a positive effect on reverberance, and background noise has a negative effect. This is confirmed in the results of the experiment. Furthermore, the objectively calculated loudness decay function provides an effective predictor of subjective decay rate, which performs better than conventional early decay time or reverberation time for the RIRs tested.     

Ambient pressure dependence of the ultra-harmonic response from contrast microbubbles

Sub-harmonic response from ultrasound contrast agent microbubbles has been demonstrated to be an effective modality for noninvasive pressure measurement. In the present study, the dependence of ultra-harmonic response on the ambient overpressure was investigated by both experimental measurements and simulations. In the measurements, the microbubbles were exposed to Gaussian pulses with varied driving frequencies and pulse lengths, at an acoustic pressure of 0.3 MPa. The amplitudes of sub- and ultra-harmonic components were measured when the ambient overpressures varied from 0-25 kPa. At the driving frequency of 1.33 MHz, the ultra-harmonic energy decreased but the sub-harmonic energy increased with the increasing overpressure; while at the driving frequency of 4 MHz, both the sub- and ultra-harmonic components showed the same tendency that the corresponding energy decreased as the overpressure was increased. A 4-MHz Gaussian pulse with 64 cycles could provide an ultra-harmonic response with both good ambient pressure sensitivity and high linearity. Furthermore, the effects of shell parameters of a microbubble on the generation of ultra- and sub-harmonic responses were discussed based on simulations using Marmottant's model. This study suggests that the ultra-harmonic response from contrast microbubbles might be applicable for noninvasive pressure measurement.     

房间脉冲响应模拟法及混响时间盲估计应用

在构建混响语声数据集时,由于缺乏真实长混响房间脉冲响应且模拟的房间脉冲响应与真实不符,因而导致数据驱动的混响时间盲估计模型性能下降。提出了一种基于条件生成对抗网络的房间脉冲响应模拟法,该方法利用真实的房间脉冲响应训练条件生成对抗网络,可以根据指定的混响时间模拟更加真实的房间脉冲响应。使用不同方法模拟的房间脉冲响应构建训练集用于训练盲估计模型,通过声学实验评估模型性能。实验结果表明,由该方法模拟的房间脉冲响应训练的估计模型在不同信噪比下均具有最小的均方根误差且在长混响情况下显著优于其他模型。     

利用高温超导直流量子干涉器件进行10$lt;sup$gt;-6$lt;/sup$gt; T量级磁场下核磁共振的研究

研究了水样品在10^-6 T量级磁场下的核磁共振谱.核磁共振信号由一个工作在液氮温度的高温超导直流量子干涉仪记录,测量在一个简易磁屏蔽室中进行.在7—70 μT的磁场范围内都观察到了15 ml水样品的核磁共振信号.相应的^1H的核磁共振频率为300—3000 Hz.在实验中获取的单次测量信噪比约为4,通过对信号的100次平均,信噪比可达到约40.进一步讨论了剩余磁场、预极化时间和采样时间对结果的影响.最后用数字滤波之后平均的方法初步得到了时域的自由感应衰减信号. 关键词： 超导量子干涉仪 核磁共振     

Estimating the mixing time of concert halls using the eXtensible Fourier Transform

In room acoustics, we measure room impulse responses (RIRs) in order to fully describe the hall. RIRs are composed of a succession of arrivals (i.e., some sound rays which have undergone one or more reflections on their way from the source to the receiver). We propose the eXtensible Fourier Transform (XFT) in order to investigate the time evolution of spectral components of RIRs. The phase evolution versus time allows to estimate the mixing time, which is defined as the time it takes for initially adjacent sound rays to spread uniformly across the room. After presenting some properties of the XFT, we show why one must compensate the natural energy decay of the RIR in order to obtain stationary signals. We estimate mixing times for a set of experimental RIRs and several that are synthesized using a stochastic model. Then, we estimate the dependance of mixing time upon the source/receiver distance in all these RIRs. Results are consistent up to the lack of reproducibility of the sound sources, but are strongly dependent on some parameters used for computing the XFT. We finally discuss the relevance of the name mixing time with respect to the theory and in regard to the time we estimate, that we propose to call cross-over time.     

Stochastic Resonance of a Periodically Driven Bistable System with Two Different Kinds of Time Delays

In this paper, we study the phenomenon of stochastic resonance (SR) in a periodically driven bistable system with correlations between multiplicative and additive white noise terms when there are two different kinds of time delays existed in the deterministic and fluctuating forces, respectively. Using the small time delay approximation and the theory of signal-to-noise ratio (SNR) in the adiabatic limit, the expression of SNR is obtained. The effects ofthe delay time τ in the deterministic force, and the delay time θ in the fluctuating force on SNR are discussed. Based on the numerical computation, it is found that: (i) There appears a reentrant transition between one peak and two peaks and then to one peak again in the curve of SNR when the value of the time delay θ is increased. (ii) SR can be realized by tuning thetime delay τ or θ with fixed noise, i.e., delay-inducedstochastic resonance (DSR) exists.     

Golay code modulation in low-power laser-ultrasound

The current work presents a correlation-based detection technique with application in modulated laser-ultrasonics. In standard use of coded sequences the impulse response of a system is recovered in the time domain with improved signal to noise ratio (SNR). The presented method is an extension of this technique, where the response to a chirped waveform is restored with improved SNR; hence, the response is in a well-defined frequency range. To achieve this goal the chirped waveforms are modulated by Golay codes. It will be shown that the response to this bandlimited carrier waveform can be recovered in the time domain with improved signal to noise ratio using a cross-correlation technique. Improvement in the SNR is discussed analytically and it is shown that this improvement is proportional to the square root of the length of the applied sequences. Experimental applications in laser-ultrasound are shown using modulated laser diodes as excitation sources with an output power of ∼1 W. In the experiments a plate with a thickness of 50 μm is investigated using Lamb waves in the MHz range to confirm the predicted improvement in the SNR. Golay codes with three different lengths were used with 7, 9 and 11 bits resulting in 2⁷ = 128, 2⁹ = 512, and 2¹¹ = 2048 repetitions in an individual signal, respectively. The predicted improvements of 2 in the SNR between the 7 and 9 bits, and between the 9 and 11 bits waveforms, respectively, were well approximated by the experimentally obtained values of 1.83 and 2.17. As Lamb wave dispersion curves can be used for the characterization of plates or layered samples by inverse problems, it is also shown that by using multiple measurement points the recovered waveforms can be utilized in the evaluation of the dispersion relation.     

Ultrasonic flaw detection using threshold modified S-transform

Interference noising originating from the ultrasonic testing defect signal seriously influences the accuracy of the signal extraction and defect location. Time–frequency analysis methods are mainly used to improve the defects detection resolution. In fact, the S-transform, a hybrid of the Short time Fourier transform (STFT) and wavelet transform (WT), has a time frequency resolution which is far from ideal. In this paper, a new modified S-transform based on thresholding technique, which offers a better time frequency resolution compared to the original S-transform is proposed. The improvement is achieved by the introduction of a new scaling rule for the Gaussian window used in S-transform. Simulation results are presented and show correct time frequency information of multiple Gaussian echoes under low signal-to-noise ratio (SNR) environment. In addition, experimental results demonstrate better and reliable detection of close echoes drowned in the noise.     

Predicted and experimental results of acoustic parameters in the new Symphony Hall in Pamplona, Spain

Two room acoustical simulation software have been used to predict the main acoustic parameters of a Symphony Hall in the planning stage, when only drawings were available. The modelled room is the Symphony Hall of the Conference Hall of Navarre, in Pamplona, Spain. Although the values of the calculation parameters (number of rays, reflection order, etc.) recommended by each software are slightly different, in this work the same values were used for both programs. Once the Hall was built, experimental results were obtained using the MLS-measurement technique. The values predicted and measured for several parameters defined in ISO 3382 at 9 receiver positions are compared. Even though the values predicted by both software are very similar for most of the acoustic parameters, there are notable differences at particular values, mainly when evaluating energy ratios. Different statistical corrections for late reflections between both programs seem to be the main reason for these differences. A more exhaustive knowledge of scattering coefficients is required to improve predictive accuracy. Important differences at 250 Hz frequency band were found between calculated and measured values probably due to the yet to be implemented seat dip effect in room simulation software. The comparison of calculated and measured impulse responses seems to be the first choice for the assessment of room simulation software. However, it should be kept in mind that its usability is also determined by many additional features. This work is not only a comparison of software dealing with the same object as well as equal input data but also shows the power of this kind of tool to predict the acoustic parameters of a room before its construction.     

Signal-to-noise ratio and frequency analysis of continuous loop averaging deconvolution (CLAD) of overlapping evoked potentials

In this study, a frequency domain formulation of continuous loop averaging deconvolution (CLAD) of overlapping evoked potentials is developed and applied for the extraction of transient responses from recordings obtained at high stimulation rates. This formulation allows for a faster execution of CLAD by using fast Fourier transform algorithms. The frequency characteristics of the deconvolution filter depends exclusively on the stimulus sequence and determines whether the noncoherent noise is amplified or attenuated in different frequencies. A formula for calculating the signal-to-noise ratio (SNR) achieved by the deconvolution process is developed. The newly developed theory and the methodology is applied to the extraction of the auditory brainstem and middle latency responses using various sequences. The effects of the sequence used and the number of sweeps averaged in ongoing acquisition on SNR are examined by using single sweep recordings. The results verify the deconvolution theory and the methodology and show its limitations. Depending on the frequency characteristics of the sequence, the deconvolution process can amplify or attenuate the EEG noise. Proper selection of the stimulus sequence can increase the SNR enhancement obtained with conventional averaging.     

一种改进的DVB-T信道估计算法

针对传统DVB-T系统信道估计算法估计精度低或复杂度高的问题,为了提高信道估计算法整体性能,提出了一种改进的DVB-T信道估计算法。该算法首先采用最小二乘法估计离散导频处频率响应估计值;进而,利用连续导频对信噪比进行估计;最后,将得到的信噪比估计值与预先设定的信噪比阈值进行比较,若估计值小于阈值,则先经过卡尔曼滤波处理,再在时频二维方向上用高斯插值得到全部子载波的频率响应值,否则直接在时域和频域上进行插值。此方法在信道条件较好时关闭卡尔曼滤波过程,从而有效减少滤波次数,提高算法整体运行速度。仿真结果表明,该方法能有效提高系统整体性能,减少在噪声影响较大时的干扰,同时也兼顾了算法整体运行速度。     

Isolating scattering resonances of an air-filled spherical shell using iterative, single-channel time reversal

Iterative, single-channel time reversal is employed to isolate backscattering resonances of an air-filled spherical shell in a frequency range of 0.5-20 kHz. Numerical simulations of free-field target scattering suggest improved isolation of the dominant target response frequency in the presence of varying levels of stochastic noise, compared to processing returns from a single transmission and also coherent averaging. To test the efficacy of the technique in a realistic littoral environment, monostatic scattering experiments are conducted in the Gulf of Mexico near Panama City, Florida. The time reversal technique is applied to returns from a hollow spherical shell target sitting proud on a sandy bottom in 14 m deep water. Distinct resonances in the scattering response of the target are isolated, depending upon the bandwidth of the sonar system utilized.