Asynchronous VLSI architecture for adaptive echo cancellation

A single chip, 128 coefficient, asynchronous echo canceller is presented. Cancellation is performed by an FIR filter whose coefficients are adapted using the power-of-two modified LMS algorithm. The pipelined circuit updates all coefficients and generates the filtered output every cycle while allowing a sampling rate >206.5 kHz     

基于相邻小波系数的图像去噪算法

针对传统的小波去噪方法在滤除噪声上的明显不足,提出了一种基于相邻小波系数的图像去噪算法。充分考虑了滤噪系数与相邻系数间的相关性,首先采用3×3像素的系数滑动窗口对噪声系数进行均值采样,然后结合双阈值法对小波系数进行缩减。实验结果表明本算法明显优于传统的滤波算法,具有较高的信噪比,是一种有效、可行的图像去噪方法。     

Efficient arbitrary sampling rate conversion with recursivecalculation of coefficients

In this paper, we present a novel algorithm for sampling rate conversion by an arbitrary factor. Theoretically, sampling rate conversion of a discrete-time (DT) sequence can be performed by converting the sequence to a series of continuous-time (CT) impulses. This series of impulses is filtered with a CT lowpass filter, and the output is then sampled at the desired rate. If the CT filter is chosen to have a rational transfer function, then this system can be simulated using a DT algorithm for which both computation and memory requirements are low. The DT implementation is comprised of a parallel structure, where each branch consists of a time-varying filter with one or two taps, followed by a fixed recursive filter operating at the output sampling rate. The coefficients of the time-varying filters are calculated recursively. This eliminates the need to store a large table of coefficients, as is commonly done     

Partial-Update L∞ -Norm Based Algorithms

The computational complexity of an adaptive filtering algorithm increases with increasing the filter tap length and therefore, the use of such a filter can become prohibitive for certain applications, especially for real-time implementation. In this paper, we develop low-complexity adaptive filtering algorithms by incorporating the concept of partial updating of the filter coefficients into the technique of finding the gradient vector in the hyperplane based on the L_infin-norm criterion. Two specific partial update algorithms based on the sequential and M-Max coefficient updating are proposed. The statistical analyses of the two algorithms are carried out, and evolution equations for the mean and mean-square of the filter coefficient misalignment as well as the stability bounds on the step size are obtained. It is shown that the proposed partial update algorithm employing the M-Max coefficient updating can achieve a convergence rate that is closest to that of the full update algorithm. Finally, simulations are carried out to validate the theoretical results and study the convergence rate of the proposed algorithms     

EBPSK解调器中利用SVM概率输出的LDPC译码

为了改善扩展的二元相移键控（EBPSK）系统在低信噪比下的误码率性能,引入了低密度奇偶校验码（LDPC）。EBPSK解调器借助特殊的冲击滤波器提高能量利用率,却增加了获得后验概率信息用于译码的困难。本文引入支持向量机（SVM）方法在滤波器输出信号中选取少量采样点进行概率输出并进行LDPC译码,仿真显示可得到较高的信噪比增益。同时,本文还仿真对比了不同采样频率及不同方式获得的后验概率信息对系统译码性能的影响,表明基于SVM的方法在低采样率和低信噪比条件下便可获得较为精确的后验概率,因此,在EBPSK系统中采用SVM方法获得后验概率信息用于LDPC译码是一种较为有效的方式。      

基于FRM结构的FIR滤波器的采样率变换方法

该文在分析滤波器传递函数的对称性与其冲激响应的关系的基础上,提出了一类具有稀疏冲激响应系数的特殊滤波器,以这类滤波器作为原型滤波器可以进一步降低FRM结构FIR滤波器的计算复杂度。并研究了基于此类FRM结构FIR滤波器的采样率变换算法、实现结构、计算复杂度及其设计问题等。最后,通过实际例子验证这种采样率变换方法的有效性。     

一种提高ADC分辨率的插值滤波方法

基于过采样和求均值原理设计插值滤波器，来提高ADC的分辨率和SNR。根据奈奎斯特采样定理，对插值滤波器进行了采样频率分析、量化误差分析，并得出分辨率与OSR的关系。通过对插值滤波器的信噪比分析，得到噪声功率与ADC位数之间的函数关系。本文对方法的有效性进行了阐述，指出插值滤波改善SNR和提高信号测量的有效位数的前提条件。实践证明可以采用插值滤波技术，通过软件实现提高仪表分辨率和信噪比，降低仪表成本。     

平坦慢衰落信道下基于HOS的PSK调制盲信道估计

提出一种基于符号高阶统计量（HOS, high-order statistics）的MPSK调制信道衰落系数盲估计算法。针对平坦慢衰落信道模型,首先分析了MPSK调制符号高阶统计量特征,证明了MPSK调制符号的M次方符号的值是唯一的,而当1≤M′相似文献   

一种有限脉冲响应滤波器格型结构优化方法及灵敏度分析

有限脉冲响应(FIR)滤波器是无线通信研究中多载波调制系统的主要组成单元.针对有限字长效应导致FIR滤波器性能下降问题,该文提出一种FIR滤波器格型结构改善因量化导致的滤波器系数误差,即降低系数灵敏度,利用状态空间结构表示相应改进格型结构系数,并推导分析其系数灵敏度表达式.仿真实例验证理论推导结果,即改进格型结构系数灵...     

Low-complexity iterative channel estimation for serially concatenated systems over frequency-nonselective Rayleigh fading channels

A low-complexity iterative maximum a posteriori (MAP) channel estimator is proposed whose complexity increases linearly with the symbol alphabet size 'M. Prediction-based MAP channel estimation is not appropriate with a high-order prediction filter or a large modulation alphabet size, since the computational complexity increases with M^L , where L is the predictor order. In contrast, the proposed channel estimator has a constant number of trellis states regardless of the prediction filter order, and is shown to provide comparable error performance to the prediction-based MAP estimator     

Polarity coincidence correlation detection of a weak noise source

One application of sample polarity coincidence correlation to the detection of a weak noise source in background noise is briefly described. Assuming an input SNR much less than one, and Gaussian input signals and noise with identical normalized power spectra, expressions for the output SNR are derived for the analog and the polarity coincidence correlator, with and without sampling. The loss in attainable SNR due to clipping and sampling is computed for three different input spectra, viz.; white noise which is passed through an RC low-pass filter, a single-tuned band-pass filter or a rectangular filter. The resulting loss is given in three diagrams, as a function of relative bandwidth of the input signal and sampling frequency. For broad-band input signals the loss is between10and1db, and between4and1db for narrow-band signals.     

12位Sigma-Delta模数转换器的降采样滤波器设计

一种由SNR（信噪比）驱动的滤波器设计,用于12位Sigma-Delta模数转换器。Sigma-Delta模数转换器包括Sigma-Delta调制器和降采样滤波器两部分,首先用Sigma-Delta调制器对信号进行过采样率量化,然后通过降采样滤波器进行数字信号处理,将信号还原到原始采样率并去除量化噪声。和传统的模数转换器相比,Sigma-Delta模数转换器具有采样率高、精度高、面积小等优点。Sigma-Delta模数转换器的滤波器设计有降采样率和滤波性能两个指标要求,该设计方法由SNR驱动并采用了两种滤波器方案,设计结果在MATLAB里进行了仿真,其SNR大于74 dB,达到12位Sigma-Delta模数转换器的要求。     

Quantizing and Sampling Considerations in Digital Phased-Locked Loops

Quantization and sampling effects on the digital phasedlocked loop (DPLL) structures obtained for demodulation of anglemodulated signals using extended Kalman filter algorithms are investigated for the high signal-to-noise ratio (SNR) case in this paper. First, the problem of quantization is considered. The validity of the uniform white sequence model for quantizer error in the DPLL is established independent of the sampling rate. Simulation results are presented for several quantizer word lengths. Also, an effective SNR is defined which allows prediction of quantized performance from unquantized results. Secondly, minimum sampling requirements for the DPLL are considered. The effect of sampling rate variation upon the predicted phase error covariance is examined. Again, simulation results are presented and compared to the predicted phase error covariance values. This results in an analytical method for determining minimum sampling rates for the DPLL. Minimum sampling rates for quantized DPLL have also been determined using the effective SNR previously defined.     

自适应线谱增强器在音频信号增强中的应用研究

本文对自适应线谱增强器（ALE）稳态的频率特性及其与输入信噪比、自适应滤波器长度和采样率的关系进行了详细分析。明确给出了AI正参数选择的原则及信号信噪比（SNR）提高的方法。在MATLAB环境下将ALE应用于音频信号增强,选择信号延迟D=100和用于自适应线性估计的滤波器长度L=32。使用块LMS=32自适应算法进行了仿真试验。仿真结果表明,AIE系统不仅提高了宽带噪声中谐波信号的信噪比．而且还具有独特的滤波功能。     

The Effects of Channel Errors in DPCM Systems and Comparison with PCM Systems

In a communications system the total system error is of importance. One measure of system error is the mean-square error between the input signal and the output signal. The total mean-squared error includes sampling error, quantization error, and channel error. The work reported here considers all three errors in differential pulsecode modulation (DPCM) systems and compares the results obtained with standard pulse-code modulation (PCM) systems. The total meansquared error is determined using well-defined system parameters such as quantizer levels, signal-to-noise ratio (SNR), sampling rate, etc. Using the derived formulas the design of DPCM systems is facilitated along with various tradeoff studies. The error equations are determined for both uniform and nonuniform quantizers. The error due to channel noise is obtained in closed form for both cases. DPCM and PCM are compared for three different reconstruction filters, the zero-order hold (ZOH), the linear interpolator (LI), and the ZOH followed by a low-pass filter. The optimum prediction coefficient is shown to depend on the channel noise. The optimum prediction coefficient improves the performance of DPCM systems considerably. DPCM is shown to perform better than PCM in all cases. Simulation results are presented, which verify the theoretical results.     

Generalised filters and optimum sampling frequencies for the reconstruction of bandlimited signals from past samples

Generalised first- and second-order filters are presented for the reconstruction of bandlimited signals. In each case the minimum sampling frequency is derived and the corresponding filter coefficients obtained. A tradeoff exists between the rate of convergence of the reconstruction and the sampling frequency.<>     

FIR filter design over discrete coefficients and least square error

The difference routing digital filter (DRDF) consists of an FIR filter followed by a first-order integrator. This structure with power-of-two coefficients has been studied as a means of achieving low complexity, high sampling rate filters which can be implemented efficiently in hardware. The optimisation of the coefficients has previously been based on a time-domain least-squares error criterion. A new design method is proposed that includes a frequency-domain least-squares criterion with arbitrary frequency weighting and an improved method for handling quantisation of the filter coefficients. Simulation studies show that the new approach yields an improvement of up to 7 dB over existing methods and that oversampling can be used to improve performance     

多目标环境中的认知雷达跟踪方法

研究了多目标环境中的认知雷达目标跟踪问题,提出了一种基于波形优化和快速粒子滤波的多目标跟踪方法。在量测模型中,基于采样的接收数据建立量测方程,以克服多目标跟踪中的数据关联问题;在状态模型中,与量测模型相匹配,联合估计目标运动状态（位置、速度）和散射系数。为实现多目标跟踪和提高跟踪性能,从联合收发自适应处理角度出发设计跟踪算法和发射波形:1）接收自适应。由于量测数据的维数以及跟踪模型的非线性程度较高,为实现对多目标的有效跟踪以及降低跟踪算法的运算复杂度,采用改进的粒子滤波方法对目标状态进行实时估计;2）发射自适应。考虑到信噪比与跟踪性能关系以及量测模型的特点,基于最优信噪比准则实现了对发射波形的优化。仿真结果表明文中所提出的跟踪方法能够有效的跟踪上目标,且所设计的自适应波形的跟踪性能优于传统固定波形。      

Optically coupled CCD transversal filter with several hundreds of coefficients at megahertz sampling rates

Under modified operating conditions, a CCD imager arrangement is used as a hybrid optoelectronic transversal filter. The principles are explained and an example of application is provided. In comparison with solutions in common use, the number of coefficients and the sampling rate are increased by several decades.     

Rayleigh商-降采样滤波器在基于调频广播信号的单站无源定位系统中的应用

根据基于调频广播信号的无源定位系统中相干脉冲压缩处理的特点,指出在相关脉冲压缩中进行降采样处理的可行性,并提出了以信噪比增益最大为评价准则设计降采样滤波器。推导了信噪比增益与降采样滤波器权系数的关系,给出了采用Rayleigh商方法求解给出准则下最优FIR滤波器的方法和步骤。仿真实验验证了在相干脉冲压缩中采用Rayleigh商-降采样滤波器既降低了计算量又保证了脉冲压缩后的信噪比增益。