共查询到17条相似文献,搜索用时 250 毫秒
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研究了RLC低通滤波器在电导受到非对称双值色噪声扰动时的随机共振现象.利用随机平均法和Shapiro-Loginov公式,得到了平均输出幅度增益的精确表达式.分析表明,在欠阻尼、临界阻尼和过阻尼RLC低通滤波器中,平均输出幅度增益对电导噪声的非对称性、相关时间、强度和信号频率都存在非单调依赖关系.适当的噪声和系统参数条件可以使有噪声时系统的平均输出幅度增益大于没有噪声时系统的平均输出幅度增益.噪声可以提高滤波器对高频信号的衰减率.恰当的条件可以获得平均输出幅度增益的最大值. 相似文献
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信号调制白噪声作用下线性系统的随机共振 总被引:1,自引:1,他引:0
研究了信号调制白噪声作用下的二阶过阻尼线性系统中的随机共振现象.基于线性系统理论和相关删去法方法,得到了系统平均输出幅度增益的精确表达式.研究表明:输出幅度增益是噪声强度,系统阻尼系数和系统固有频率,以及激励信号频率的非单调函数,适当的噪声强度和系统参数可以使噪声情况下的输出幅度增益大于无噪声时的输出幅度增益. 相似文献
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《二值噪声作用下线性系统的随机共振》 总被引:2,自引:0,他引:2
研究了二值噪声用下的二阶过阻尼线性系统的随机共振现象。基于线性系统理论和相关删去法方法,得到了系统平均输出幅度增益的精确表达式。研究表明:输出幅度增益是噪声的强度和相关时间、系统阻尼系数,以及激励信号的频率的非单调函数;另外,适当的噪声参数和系统参数可以使噪声情况下的输出幅度增益大于无噪声时的输出幅度增益。 相似文献
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采用四阶Runge-Kutta算法,以信噪比增益和谱功率放大率为随机共振测度指标,研究了电容参数和激励信号均受高斯白噪声扰动时RLC串联谐振电路的随机共振现象。研究表明,欠阻尼RLC串联谐振电路在适当的电路参数、激励信号频率和噪声参数条件下,信噪比增益和谱功率放大率是噪声强度和输入信号频率的非单调函数,且其值均大于1,电路中存在随机共振现象;而在临界阻尼或过阻尼情况下尽管信噪比增益在一定条件下也是噪声强度和输入信号频率的非单调函数,但谱功率放大率取值小于1,电路发生了随机共振现象,改善了信噪比,信号能量却并未得到加强。这一结论使得将欠阻尼RLC串联谐振电路应用于微弱信号检测成为可能。 相似文献
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以经典力学模型——质量-阻尼-弹簧欠阻尼二阶线性系统为研究对象,当系统的阻尼系数和固有频率同时受乘性高斯噪声干扰时,利用此系统产生的随机共振来消除此类噪声.理论分析表明,欠阻尼二阶线性系统中存在随机共振现象,系统的平均输出幅度增益呈现非单调变化,不仅在一定条件下大于无噪声时的增益,而且调节适当的系统参数和噪声强度能够提高幅度增益.因此,采用可视化仿真软件SIMULINK建立仿真模型,并进行实例模拟.仿真进一步表明,通过调节适当的系统参数或噪声强度,使系统处于共振区域,就会把夹杂在噪声中的被测信号突现出来,从而实现了弱信号的检测,证实该方法消除乘性噪声的可行性和有效性. 相似文献
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基于线性系统理论,利用三值噪声的统计性质,推导出了系统输出幅度增益的解析表达式。研究表明,随着输入信号频率、三值噪声的定态概率、噪声状态间的跃迁速率、参数k(电容C的倒数)的增大,输出幅度增益出现一个最大值。当信号频率较低时,输出幅度增益随着k的增大而单调下降;当信号频率较高时,输出幅度增益随着k的增大而单调增大。当信号频率较低时,输出幅度增益随着定态概率的增大而单调增大;当信号频率较高时,输出幅度增益随着定态概率的增大而单调下降。随着G0(电阻的倒数)的增大,输出幅度增益单调减小。 相似文献
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《振动与冲击》2021,(3)
针对实际工程应用中系统的非对称性,将α稳定分布噪声与非对称三稳系统相结合,以平均信噪比增益(A-SNRI)为性能指标,分别探究了α稳定分布噪声的特征指数α和对称参数β取不同值时,系统参数a、b,刻画三稳系统非对称性的偏度r以及噪声强度放大系数D对非对称三稳系统随机共振输出的规律。研究结果表明:在α稳定分布噪声环境下,通过适当调节系统结构参数a、b和偏度r均可诱导随机共振,实现对微弱信号的检测,并且性能较好区间不随α或β的变化而改变;在研究噪声诱导的随机共振现象时,调节噪声强度放大系数D也可诱导随机共振,且共振效果较好的D区间也不随α或β的变化而变化。这些研究结论为α稳定分布噪声下非对称三稳系统共振中参数的合理选取提供了理论依据。 相似文献
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研究了在非对称双值噪声、白噪声和方波共同作用下受限系统中的熵随机共振现象。在绝热近似条件下,推导出了输出信噪比的表达式。分析表明,信噪比是非对称双值噪声的强度与非对称参数、白噪声强度,以及方波信号幅度的非单调函数。另外,信噪比随受限结构参数的变化而非单调变化。讨论了双值噪声的相关率对SNR的影响。 相似文献
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The linearity of an efficient polar transmitter architecture, with a 1 bit oversampled delta?sigma (DS) modulating the envelope signal, depends, to a high degree, on low-pass envelope filtering. This filter is compulsory to attenuate the DS quantisation noise. A high cut-off frequency results in more noise being included. In contrast, using a filter with a low cut-off frequency results in attenuation of the information content of the envelope signal. Either way, the result is unwanted spectral regrowth. By pre-emphasising the envelope signal, the filter?s attenuation of the information is mitigated. The pre-emphasis is implemented by a digital pseudo-derivative high-pass filter, with inverse magnitude characteristics of the analogue low-pass filter, within a limited interest band. Consequently, the low-pass filter can be designed with a lower cut-off frequency to attenuate more of the DS modulator noise, and the modulator can switch at lower frequencies. With this technique, the WLAN output spectrum, at the critical 30 MHz offset corner frequency, is improved by 12.5 dB, considering a second order DS sampling at 1.28 GHz. The technique was verified with an experimental setup and the behaviour agrees well with simulations. 相似文献
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Adaptive IIR filter design for single sensor applications 总被引:2,自引:0,他引:2
The goal of this research was to investigate the theoretical design and physical implementation of a digital adaptive IIR filter to serve as an enhancement to the traditional active RC or passive RLC anti-aliasing filter. This all-digital filter will reside directly on the DSP engine. As explained in the paper, the adaptive IIR filter is designed to process an oversampled signal coming from a single sensor to reject noise in an acquisition system. Differentiation between the noise and the signal is obtained by exploiting the different auto-correlation functions of the two signals. In contrast to oversampling techniques employed in sampled data systems that are designed to relax the requirements of an analog anti-aliasing filter, this filter will track a signal in the frequency domain. Several power spectral density plots are given to illustrate the performance of the new filter. The results also indicate that the new filter performs well as compared to the Wiener filter in the stationary case 相似文献
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Spectral histogram using the minimization algorithm-theory and applications to flaw detection 总被引:3,自引:0,他引:3
Li X Bilgutay NM Murthy R 《IEEE transactions on ultrasonics, ferroelectrics, and frequency control》1992,39(2):279-284
In ultrasonic flaw detection in large grained materials, backscattered grain noise often masks the flaw signal. To enhance the flaw visibility, a frequency diverse statistical filtering technique known as split-spectrum processing has been developed. This technique splits the received wideband signal into an ensemble of narrowband signals exhibiting different signal-to-noise ratios (SNR). Using a minimization algorithm, SNR enhancement can be obtained at the output. The nonlinear properties of the frequency diverse statistic filter are characterized based on the spectral histogram, which is the statistical distribution of the spectral windows selected by the minimization algorithm. The theoretical analysis indicates that the spectral histogram is similar in nature to the Wiener filter transfer function. Therefore, the optimal filter frequency region can be determined adaptively based on the spectral histogram without prior knowledge of the signal and noise spectra. 相似文献
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Improving the response of a wheel speed sensor by using frequency-domain adaptive filtering 总被引:1,自引:0,他引:1
In this paper, a frequency-domain least-mean-square adaptive filter is used to cancel noise in a wheel speed sensor embedded in a car under performance tests. In this case the relevant signal is buried in a broad-band noise background, where we have little or no prior knowledge of the signal or noise characteristics. The results of the experiments show that the signal of interest and the noise (all forms of interference, deterministic, as well as stochastic) share the same frequency band and that the filter used significantly reduced the noise corrupting the information from the sensor while it left the true signal unchanged from a practical point of view. In this paper, a signal-to-noise ratio improvement higher than 40 dB is achieved. The results of the experiment show the importance of using digital signal processing when dealing with a signal corrupted by noise. 相似文献
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船舶辐射噪声的包络谱中蕴含着轴频和桨叶数等船舶固有特征信息,对船舶目标识别具有重要意义。为了提高船舶辐射噪声包络谱解调性能,提出了基于变分模态分解(Variational Mode Decomposition, VMD)和窄带包络相关的改进DEMON分析方法。首先利用VMD算法代替传统带通滤波器,将船舶辐射噪声信号分解为若干个子带;然后对各子带进行希尔伯特(Hilbert)检波并计算平均窄带包络相关系数,用于衡量信号的包络调制在频域上的非均匀性;最后提取各子带信号包络谱并按照平均窄带包络相关系数进行加权融合,从而得出宽带噪声信号的包络谱。利用该方法对实测不同类型和不同航速船舶辐射噪声信号进行了处理,结果均表明所提方法能有效提高包络谱解调效果,较传统方法更为有效。 相似文献
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Limitations on accuracy of Doppler estimation in continuous-wave noise radar with correlation processing are studied. Second-order properties of output of the correlation receiver are evaluated and an approximate Cramer-Rao bound on errors of Doppler measurement is derived. The accuracy of Doppler measurements is found to be affected by the following factors: power spectral density of noise signal, frequency response of the low-pass filter in correlator, observation time, velocity of the target and signal to noise ratio. It is shown that the random nature of the transmitted signal induces additional fluctuations at the output of correlator which limit the accuracy even in the infinite signal to noise case. Qualitative extension of the results to a case covering multiple targets and clutter is made. It is argued that the performance will decrease and that increasing transmitted power may not provide significant improvement when clutter is present. 相似文献
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Planat M Eckert C 《IEEE transactions on ultrasonics, ferroelectrics, and frequency control》2000,47(5):1173-1182
A mixer cascaded with a low-pass filter is the basic piece of any communication receiver. The frequency and amplitude fluctuations of beat signals recorded at the output of the mixer plus filter set-up are investigated experimentally and explained on the basis of number theory in relation to the Riemann problem concerning the distribution of prime numbers. The frequency of the beat signal is obtained from a diophantine approximation of the frequency ratio of the input oscillators. The amplitude is defined globally from the position of resolved fractions with respect to an equally spaced graduation. Time series analysis methods show that frequency fluctuations present a transition from white frequency noise to 1/f frequency noise close to resonance; the latter is compatible with an underlying fractal attractor. The same transition is observed in the case of time series computed from continued fraction expansions. 相似文献