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1.
Harmonic decomposition of audio signals with matching pursuit   总被引:3,自引:0,他引:3  
We introduce a dictionary of elementary waveforms, called harmonic atoms, that extends the Gabor dictionary and fits well the natural harmonic structures of audio signals. By modifying the "standard" matching pursuit, we define a new pursuit along with a fast algorithm, namely, the fast harmonic matching pursuit, to approximate N-dimensional audio signals with a linear combination of M harmonic atoms. Our algorithm has a computational complexity of O(MKN), where K is the number of partials in a given harmonic atom. The decomposition method is demonstrated on musical recordings, and we describe a simple note detection algorithm that shows how one could use a harmonic matching pursuit to detect notes even in difficult situations, e.g., very different note durations, lots of reverberation, and overlapping notes.  相似文献   

2.
An approach for reconstructing the input currents in a cable current transformer (CCT) through the output signals is reported. A PSpice circuit simulator is used for the analysis of the transients in the equivalent nonlinear circuit. Such a procedure is very useful in the design of selective earth-fault protections and in many cases when the input excitations are to be found by known reactions  相似文献   

3.
周鑫  金星 《红外与激光工程》2014,43(6):1722-1727
波长调制谱技术(WMS)在慢扫描信号基础上叠加上高频的调制电流,并加到半导体激光器的驱动电流上,于是激光频率在线性扫描的同时受交流调制。频率调制的激光束在通过吸收气体以后,其吸收线的强度也受到相同频率调制。采用谐波小波方法作为数据解调的手段,对WMS技术产生的信号进行处理并实现了2f信号的提取。采用这种方法对波数为7185.6 cm-1的水蒸气谱线进行测量实验,调制频率达到180kHz,对数据进行谐波小波处理后获得了高质量的2f信号,该2f 信号的峰值中心辨识度高,其数据光滑度、信号的抗干扰能力方面均优于数字锁相方法。  相似文献   

4.
The proposed nonlinear finite impulse response (FIR) differentiator is optimised for the reduction of quantisation error when processing signals of constant or slowly varying rate. Judicious use of a number theoretic approach to rate estimation for some input sequences reduces mean-squared error, relative to that of the optimum linear differentiator, whilst facilitating simple implementation.  相似文献   

5.
Harmonic signals reflected from electroabsorption modulators are measured and analyzed as a function of modulator bias. The second-harmonic signal exhibits a dip close to the bias where the maximum of RF link gain occurs, over a significant optical power range. Using an equivalent circuit analysis we show this is caused by the inherent electroabsorption effect. The second-harmonic signal can be exploited for dynamic self-bias control of electroabsorption modulators in analog fiber-optic links  相似文献   

6.
Different structures of universal and automated integral monitoring systems (AIMS) for broadcasting are discussed. A procedure determining the probability characteristics of broadcasting receivers input signal levels is investigated, and useful statistic distribution laws are obtained. The results enable us to propose an integral quality estimation for an amplitude modulated broadcasting system operating in the 0.15 to 30 MHz frequency range. This method can be applied to various modulation methods and frequency ranges  相似文献   

7.
This paper proposes a novel proportionate normalized least‐mean‐squares (PNLMS) algorithm that is robust to input noises. Through compensating for biases due to input noise added at the filter input, the proposed PNLMS algorithm avoids performance deterioration owing to the noisy input signals. Moreover, since the proposed PNLMS algorithm uses a new gain‐distribution matrix, it has a fast convergence rate compared with the existing PNLMS algorithms, even when there is no input noise. The experimental results verify that the proposed PNLMS algorithm enhances the filter performance for sparse system identification in the presence of input noises.  相似文献   

8.
《现代电子技术》2015,(11):95-97
在刚性转子动平衡修正中,对有效的振动信号进行检测和提取是直接关系到后期计算与动平衡修正效果好坏的最关键的一个环节,谐波小波具有良好的盒形谱特性、存在明确的函数表达式、时域定位准确和算法实现简单等优点,而在谐波小波的基础上实现的谐波小波包算法既拥有谐波小波的优点,并且还能对分析频段进行任意的划分,以便提取和分析更加细致的信息,通过模拟给出了采样信号,并使用谐波小波包算法对采样信号进行滤波处理,有效地滤除了干扰信号,最终成功地提取到了目标振动信号。  相似文献   

9.
Consider a channel modeled as a linear operator with additive noise that arises from independent samples of the input. Sufficient conditions are presented for the existence of anm-dimensional input signal that can be recovered by means of a linear operator on the channel output.  相似文献   

10.
《Signal processing》1987,12(2):169-176
It is known from literature how to design optimal quantizers for input signals with a given probability density function. In this paper, optimal quantizers for input signals corrupted by noise are designed. In addition to that it is shown by a separation theorem that the optimal quantizer can be replaced by an estimator for the corrupted input signal followed by an optimal quantizer for the estimate. Furthermore, some numerical results for processes with a Gaussian probability density function are given.  相似文献   

11.
The analysis of a periodic signal with localized random (or high frequency) noise is given by using harmonic wavelets. Since they are orthogonal to the Fourier basis, by defining a projection wavelet operator the signal is automatically decomposed into the localized pulse and the periodic function. An application to the analysis of a self-similar non-stationary noise is also given.  相似文献   

12.
To design broadband matching networks for microwave communication systems, commercially available computer aided design (CAD) tools are always preferred. But these tools need proper matching network topology and element values. Therefore, in this paper, a practical method is proposed to generate distributed-element matching networks with good initial element values. Then, the gain performance of the designed matching network can be optimized employing these tools. The utilization of the proposed method is illustrated by means of the given example. It is shown that proposed method provides very good initials for CAD tools.  相似文献   

13.
This paper presents an algorithm that adapts the parameters of a Hammerstein system model. Hammerstein systems are nonlinear systems that contain a static nonlinearity cascaded with a linear system. In this paper, the static nonlinearity is modeled using a polynomial system, and the linear filter that follows the nonlinearity is an infinite-impulse response (IIR) system. The adaptation of the nonlinear components is improved by orthogonalizing the inputs to the coefficients of the polynomial system. The step sizes associated with the recursive components are constrained in such a way as to guarantee bounded-input bounded-output (BIBO) stability of the overall system. This paper also presents experimental results that show that the algorithm performs well in a variety of operating environments, exhibiting stability and global convergence of the algorithm.  相似文献   

14.
A new algorithm is proposed for the deconvolution of an unknown, possibly colored, Gaussian or nonstationary signal that is observed through two or more unknown channels described by rational system transfer functions. More specifically, not only the root (pole and zero) locations but also the orders of the channel transfer functions are unknown. It is assumed that the channel orders may be overestimated. The proposed algorithm estimates the orders and root locations of the channel transfer functions, therefore it can also be used in multichannel system identification problems. The input signal is allowed to be nonstationary and the channel transfer functions may be a nonminimum phase as well as noncausal, hence the proposed algorithm is particularly suitable for applications such as dereverberation of speech signals recorded through multiple microphones. Several experimental results indicate improvement compared to the existing methods in the literature  相似文献   

15.
Many bioelectric signals result from the electrical response of physiological systems to an impulse that can be internal (ECG signals) or external (evoked potentials). In this paper an adaptive impulse correlated filter (AICF) for event-related signals that are time-locked to a stimulus is presented. This filter estimates the deterministic component of the signal and removes the noise uncorrelated with the stimulus, even if this noise is colored, as in the case of evoked potentials. The filter needs two inputs: the signal (primary input) and an impulse correlated with the deterministic component (reference input). We use the LMS algorithm to adjust the weights in the adaptive process. First, we show that the AICF is equivalent to exponentially weighted averaging (EWA) when using the LMS algorithm. A quantitative analysis of the signal-to-noise ratio improvement, convergence, and misadjustment error is presented. A comparison of the AICF with ensemble averaging (EA) and moving window averaging (MWA) techniques is also presented. The adaptive filter is applied to real high-resolution ECG signals and time-varying somatosensory evoked potentials.  相似文献   

16.
在本文中讨论了公度线网络的频域和时域响应间的关系,并提出用时域化方法设计公度线网络,可以提高机助设计的计算效率。给出若干实例,说明时方法有助于实际微波电路元件的设计。  相似文献   

17.
An efficient harmonic-balance technique for determining the steady-state response of nonlinear circuits to multitone input signals is presented. Its main feature is an approach for calculating the Jacobian matrix associated with the Newton Raphson algorithm which significantly reduces its computational cost.  相似文献   

18.
Expressions for calculating current derivatives efficiently with respect to voltage in nonlinear conductances under periodic excitation are derived. The validity of Egami's approach for computing these derivatives is discussed.  相似文献   

19.
Ali Selk Ghafari  Mehdi Behzad   《Mechatronics》2005,15(10):1175-1189
This paper gives the results of simulation and experimental investigation on the effects of random signals on the accuracy of micro-stepping control positioning. For studying and simulation of the effect of random noise signals on performance of the accurate position control systems, such as Hybrid Stepper Motors (HSMs), a micro-step driver and controlling unit using PID controller has been designed and constructed. Several parametric studies have been carried out including different white noise power and micro-step per revolution. Tracking problem for a HSM model has been simulated, and the experimental study for similar cases has been carried out by implementing the designed controller in real-time operation by using Real Time Windows Target Toolbox of Matlab software and Simulink. Simulation and experimental results show that random noise source changes current profile and affects the accuracy of positioning. Performance of the proposed PID controller under the implementation of random noise on phases one and two of stepper motor has been proved to be accurate enough even under disturbance load currents, on the system. Experimental and simulation results show the good performance of designed controller in tracking problem, affected by various random noise powers and motor speeds in different micro-step positions. Moreover there is an excellent agreement between experimental and simulation results.  相似文献   

20.
Bufferless distributed circuit (BDC) broadcasting is proposed as a technique for broadcasting high-speed chip input signals to a series of on-chip destination cells as needed in crosspoint switch, parallel multiplier, distributed amplifier, etc., chip designs. In contrast with conventional techniques that use an on-chip buffer to assist broadcasting, BDC broadcasting offers the advantage of lower signal delay and power dissipation. In an experimental GaAs heterojunction bipolar transistor (HBT) 8×4 crosspoint switch assembly, BDC broadcasting was found to achieve a 40% power savings with little or no penalty in jitter or bit error rate performance at a 10-Gb/s data rate  相似文献   

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