Accurate Extraction of Noise Source Impedance of an SMPS Under Operating Conditions

An accurate measurement method to extract the common mode (CM) and the differential mode (DM) noise source impedances of a switched-mode power supply (SMPS) under its operating condition is developed and validated. With a proper premeasurement calibration process, the proposed method allows extraction of both the CM and the DM noise source impedances with very good accuracy. These noise source impedances come in handy to design an electromagnetic interference filter for an SMPS systematically with minimum hassle.      

一种具有可重构陷波功能的差分宽带带通滤波器

文章提出了一种具有陷波可重构功能的差分宽带带通滤波器,具有良好的差模响应与共模抑制效果。所设计的差分带通滤波器通过采用对称的四阶分支线结构,在差模激励下可等效出电壁,在共模激励下可等效出磁壁。同时,该滤波器通过对称地耦合两个1/4波长的阶梯阻抗谐振器来产生所需频段的陷波特性,并通过改变阶梯阻抗谐振器上变容二极管两端的直流偏置电压来改变阶梯阻抗谐振器的电长度,从而调整陷波的频段。仿真和测试结果表明该差分带通滤波器的工作频带为2.7~7.3 GHz,相对阻抗带宽为92%。在工作频段中,差模回波损耗均大于10 dB,共模抑制大于15 dB。随着变容二极管两端直流控制电压从10.3 V变化到3.6 V,陷波的中心频率从5.6 GHz移动到6.1 GHz,同时滤波器宽带带通特性基本保持不变。     

开关电源EMI滤波器设计

分析了一种典型的开关电源电路,利用Pspice软件对其传导电磁干扰进行仿真研究,以TDK公司提供的元器件模型,提出了一种二阶无源EMI滤波器,完全消除了电路输出信号中的尖峰干扰,抑制了开关电源电路中的共模、差模噪声。同时,研究源和负载理想、非理想阻抗特性对滤波器插入损耗的影响,具有一定的意义。     

基于SRAM和FPGA的DSO深存储功能设计

存储深度决定了数字存储示波器能够连续采集信号的最大时长,也决定了示波器在各个时基档位的实时采样率,提高存储深度有助于提高示波器的连续捕获时间和实时采样率。文章采用SRAM进行数据存储,利用FPC．A接收采样率为1Gsps的ADC的采样数据及控制SRAM的读写等,实现了采样率为1Gsps、存储深度为IM的深存储功能设计。     

A programmable transversal filter for voice-frequency applications

A fully integrated, programmable transversal filter optimized for low-noise, low-power, voice-frequency applications is described. The filter, fabricated with a standard double-poly NMOS process, achieves convolution of an analog input signal with digital tap weightings using a structure with sample-and-hold gates for analog storage and a multiplexed MDAC for multiplication. The design of the filter eliminates fixed pattern noise usually associated with such structures and enables a dynamic range in excess of 70 dB (LPF, f/SUB o//f/SUB s/=0.08) to be achieved at an 8 kHz sampling rate with a power dissipation of less than 80 mW. This area efficient device forms the basis for a range of possible voice-band signal processing functions.     

一种非均匀采样下小信号的检测方法

非均匀采样由于其具有不受采样频率限制、频率分辨率高以及抗混叠等优点,使得其应用十分广泛。但非均匀采样会引起信号的频谱噪声,这样使得非均匀采样下小信号的检测不易实现。本文分析了非均匀采样引起频谱噪声的原因,提出一种基于非均匀采样的小信号检测方法。该方法根据非均匀采样检测得到的大幅度信号,应用陷波器将其消除,降低了由大信号引起的频谱噪声,从而检测出小信号。文中详细说明了陷波方法的原理、陷波器宽度和深度的选择、陷波器中心频率的确定以及陷波器在非均匀采样下的应用,最后给出实验结果。理论和实验表明,基于非均匀采样的陷波方法是一种行之有效的信号频率检测方法,使用该方法处理信号可以得到准确的频率估计效果,检测出信号幅度相差100倍以上的多个信号频率。     

Optimal filter banks for signal reconstruction from noisy subbandcomponents

Conventional design techniques for analysis and synthesis filters in subband processing applications guarantee perfect reconstruction of the original signal from its subband components. The resulting filters, however, lose their optimality when additive noise due, for example, to signal quantization, disturbs the subband sequences. We propose filter design techniques that minimize the reconstruction mean squared error (MSE) taking into account the second order statistics of signals and noise in the case of either stochastic or deterministic signals. A novel recursive, pseudo-adaptive algorithm is proposed for efficient design of these filters. Analysis and derivations are extended to 2-D signals and filters using powerful Kronecker product notation. A prototype application of the proposed ideas in subband coding is presented. Simulations illustrate the superior performance of the proposed filter banks versus conventional perfect reconstruction filters in the presence of additive subband noise     

基于补偿原理的逆变器EMI抑制方法研究

由于逆变器电源对输出接地电容的限制,对严重的共模电磁干扰难以用大容量的共模滤波器抑制共模噪声。文章介绍了一种新型的基于噪声电流补偿原理的电力电子装置无源干扰抑制技术,探讨了不同方案实现对共模噪声电流的补偿,并在一台DC/AC逆变器中进行了补偿测试。实验结果证实了该EMI抑制方法的有效性。     

Signal processing in adaptive arrays using power basis

The development of the theory of adaptive arrays (AAs) is proposed based on the representation of the inverse covariance matrix (CM) of a noise in the AA channels as a finite power series expansion using the direct CM and by representation of a weight vector of the AA as a finite series expansion of the power vectors. The dimension of the power CM basis is equal to the power of the minimum polynomial of the CM. In the case when the number of external interference sources is less than the number of AA channels, such polynomials have the same fundamental role as the characteristic polynomial of the CM in an opposite case. Proofs for the existence of the above mentioned polynomials of the CM are given. A new method for the calculation of the polynomial coefficients is presented, and the physical properties of the power vector basis are studied. It is shown that the power vectors are correlated and that there are two stages of AA signal processing.     

一种2Gsps数字示波器数据采集系统的设计

数据采集系统是数字存储示波器的核心部分。在研究剖析数字存储示波器工作原理的基础上,围绕其数字系统的设计与实现,采用ADC＋高频时钟电路＋FPGA＋DSP的结构模式,实现最高实时采样率2Gsps、分辨力8bit、存储深度8MB／CH的示波器数据采集系统,为国内高速数据采集系统设计提供了一种解决方案。     

Series/Parallel Time-Multiplexed Switched-Capacitor Filters with Programmability Based on Non-Uniform Sampling

This paper presents an improved scheme for programmable time-multiplexed (TM) switched-capacitor (SC) filters. The proposed approach uses a novel sampling technique, which eliminates the need for resolution/area tradeoffs. The programmability of each processing channel is based on the use of non-uniform clock signals with noise-shaped sampling energy. No capacitor values are modified for programming frequency response parameters and, hence, the performance of the TM SC filter is not sacrificed for programmability. Such a sampling technique not only leads to an accurate frequency response control, but also allows the design procedures and the resulting SC circuit implementation to be simplified. A test-chip including a programmable second-order TM SC filter with a multiplexing order of four, which operates in series or in parallel mode, was fabricated in conventional CMOS technology. Measurement results demonstrate the effectiveness and versatility of the proposed technique.     

基于AT89S52和K9F6408UOA的语音数字系统设计

在研究传统语音录放电路的基础上,提出了一种基于AT89S52的音频信号采集、存储与处理系统。该系统以单片机AT89S52为控制器,采用键盘和LCD作为人机界面,ADC0809采集音频信号,扩展8MB闪速存储器K9F6408UOA作为数字化音频信号的存储器,通过软件滤波滤除噪音;采用PWM产生声音的原理,使存储在Flash中的音频数据控制PWM每个波形的占空比,通过低通滤波器将声音从PWM的脉冲中分离,并驱动扬声器。实验表明：8kHz采样频率和8位采样位数可获得清晰的语音以及较好的音乐声,语音存储时间达15min。     

传导EMI的测试

在传导EMI(电磁干扰)中既存在共模电流也存在差模电流,要很好的分析传导EMI系统,就很有必要对共模干扰信号和差模干扰信号分离开来单独进行测试.这样,共模和差模噪声的分离就显得很重要了.     

Predictive interpolation and decimation of narrow-band signals

A digital filter design and optimization method for sampling rate alteration of narrow-band signals is presented. The resulting predictive finite-duration impulse-response filters are used to remove the aliasing or imaging frequency components when the sampling rate is changed. The sampling rate can be changed either by an integer factor or a rational factor. A least-squares-based generalization is developed for optimal wide-band noise attenuation     

Adaptive robust impulse noise filtering

It is well known that when data is contaminated by non-Gaussian noise, conventional linear systems may perform poorly. The paper presents an adaptive robust filter (adaptive preprocessor) for canceling impulsive components when the nominal process (or background noise) is a correlated, possibly nonstationary, Gaussian process. The proposed preprocessor does not require iterative and/or batch processing or prior knowledge about the nominal Gaussian process; consequently, it can be implemented in real time and adapt to changes in the environment. Based on simulation results, the proposed adaptive preprocessor shows superior performances over presently available techniques for cleaning impulse noise. Using the proposed adaptive preprocessor to clean the impulsive components in received data samples, conventional linear systems based on the Gaussian assumption can work in an impulsive environment with little if any modification. The technique is applicable to a wide range of problems, such as detection, power spectral estimation, and jamming or clutter suppression in impulsive environments     

全相位滤波白噪声产生有色噪声及其功率谱估计

本文提出用全相位数字滤波器这种新型滤波器滤波白噪声产生有色噪声的方法,这种方法比用传统FIR滤波器产生的有色噪声频谱泄漏少,阻带衰减功率大等优点.并且分别用Welch法和全相位法对在有色噪声中混入的双频小信号余弦波进行谱估计,实验证明全相位谱估计性能优于经典谱估计.     

Optimum sampling vectors for Wiener filter noise reduction

Sampling is a very important and basic technique for signal processing. In the case that noise is added to a signal in the sampling process, we may use a reconstruction and noise reduction filter such as the Wiener filter. The Wiener filter provides a restored signal of which the mean square error is minimized. However, the mean square error by the Wiener filter depends on the sampling vectors. We may have the freedom to construct sampling vectors. We provide optimum sampling vectors under the condition that the Wiener filter is used for noise reduction for two cases wherein the noise is added before/after sampling. The sampling vectors provided in this paper may not be practical since they are very complicated. However, the minimum mean square error, which we provide theoretically, can be used for evaluating other sampling vectors. We provide all proofs of the theorems and lemmas. Furthermore, by experimental results, we show their advantages     

Mixed H2/H∞ filtering design inmultirate transmultiplexer systems: LMI approach

A mixed H₂/H_∞ filter design is proposed for multirate transmultiplexer systems with dispersive channel and additive noise. First, a multirate state-space representation is introduced for the transmultiplexer with the consideration of channel dispersion. Then, the problem of signal reconstruction can be regarded as a state estimation problem. In order to design an efficient separating filterbank for a transmultiplexer system with uncertain input signal and additive noise, the H_∞ filter is employed for robust signal reconstruction. The H₂ filter design is considered to be a suboptimal approach to achieve the optimal signal reconstruction in transmultiplexer system under unitary noise power. Finally, a mixed H₂/H_∞ filter is proposed to achieve a better signal reconstruction performance in transmultiplexer systems. These design problems can be transformed to solving the eigenvalue problems (EVP) under some linear matrix inequality (LMI) constraint. The LMI Matlab toolbox can be applied to efficiently solve the EVP by convex optimization technique     

Estimation of Aliasing Effects Due to Periodical Nonuniform Individual Sampling in High-Q Switched-Capacitor Filters

The periodical nonuniform individual sampling scheme has been shown suitable for capacitance spread and total capacitor area reduction in high quality (Q) factor switched-capacitor (SC) filters. However, the use of periodical nonuniform clock signals results in additional aliasing components in the output spectrum. This paper presents a simple model to analyze the generation of such alias components and gives practical expressions to estimate their power. The results are verified through circuit simulation of a 10.7-MHz second-order SC bandpass filter in a 0.35-mum CMOS technology. Implications on the use of this technique in the design of intermediate-frequency filters are discussed     

Design and analysis of a CMOS passive Σ∆ ADC for low power RF transceivers

Design and analysis of a Σ∆ modulator with a passive switched capacitor loop filter is presented. Design steps for optimum loop filter design for quantization noise suppression and thermal noise reduction is outlined. Design specifications for sampling clock phase noise, reference buffer and input buffer settling is analyzed. Presented design has a 2nd-order loop filter and uses only metal-metal capacitors and thin oxide digital transistors with no additional components occupying less than 0.1 mm² silicon area in 0.13 μm CMOS digital process. Measurement results show that the ADC achieves 80 dB peak SNR at a 100 kHz integration bandwidth with 1 pJ/sample conversion efficiency. With decimation filter power consumption of 0.22 mW at 104 MHz sampling rate, the ADC consumes only about 1 mA at 1.5 V for each channel.