采用数字滤波器的音频系统的频域均衡(二)──均衡实例

４．２ＦＩＲ部分数字反折叠和重现滤波器——根据实测频响进行设计部分数字反折叠和重现ＦＩＲ滤波器被设计成用于时域和频域中的线性系统参数建模应用的音频数据采集和测试信号发生的组成部分。４．１节讨论了设计规范。ＤＳＰ的取样频率为ｆＴ／４＝５０ｋＨｚ，对应于...

基于MATLAB的数字滤波器设计

数字滤波器是数字信号处理中重要组成部分,它的高精度、绝对的可靠性,使其具有很多优点。根据模拟过滤器的设计原理,本文提出数字滤器设计办法,优化最佳频率办法,通过分析制定相应频率特征,设计的滤波器符合技术指标。

Design and Implementation of a Novel Area-Efficient Interpolator

This paper presents the design considerations and implementation of an area-efficient interpolator suitable for a delta-sigma D/A converter. In an effort to reduce the area and design complexity, a method for designing an FIR filter as a tapped cascaded interconnection of identical subfilters is modified. The proposed subfilter structure further minimizes the arithmetic number. Experimental results show that the proposed interpolator achieves the design specification,exhibiting high performance and hardware efficiency,and also has good noise rejection capability. The interpolation filter can be applied to a delta-sigma DAC and is fully functional.

激光引信电流采样处理系统设计与应用

激光引信是一种对目标的高精度主动检测方案,广泛应用于空空导弹中。提出一种对激光引信电流采样信号进行调理和数字信号处理相结合的方案,处理后的信号将可用于调节激光器供电电压,以实现激光器恒流驱动的目的。为分析此方法的性能,给出了精确分析模型。结果表明,与传统的激光引信电源采样处理方式相比,对信号采样处理系统的硬件和软件进行优化设计,能够对采样的信号噪声幅值降低50%,可减弱激光引信发射功率抖动,大大提高激光引信的稳定性和精度。该处理系统成功应用于某型激光引信发射驱动电路中,经充分考核,验证了其有效性。

A 16bit 96kHz Chopper-Stabilized Sigma-Delta ADC

A 16bit sigma-delta audio analog-to-digital converter is developed.It consists of an analog modulator and a digital decimator.A standard 2-order single-loop architecture is employed in the modulator.Chopper stabilization is applied to the first integrator to eliminate the 1/f noise.A low-power,area-efficient decimator is used,which includes a poly-phase comb-filter and a wave-digital-filter.The converter achieves a 92dB dynamic range over the 96kHz audio band.This single chip occupies 2.68mm2 in a 0.18μm six-metal CMOS process and dissipates only 15.5mW power.

Demodulation System for Fiber Bragg Grating Sensors Using Digital Filtering Technique

A discrimination measurement method and demodulation technique for fiber Bragg grating (FBG) sensors were presented using digital filtering technique. The system can control a tunable fiber Fabry-Perot filter with sawtooth wave voltage generated by digital clock to interrogate FBG sensors. Using the analogue digital converter (ADC), the reflected FBG signals were sampled with synchronous digital clock. With the aid of digital matched filtering technique, the sampled FBG signals were processed to obtain the maximum signal-to-noise ratio (SNR) and the Bragg wavelength shift from the FBG signals was recovered. The results demonstrate that this system has a scanning range of 1 520 nm-1 575 nm,and the wavelength detection accuracy is less than 2 pm with 1.5 Hz scanning frequency.

Development of typhoon driven wave nowcasting model in Southeast China Sea

Using optimal interpolation data assimilation of observed wave spectrum around Northeast coast of Taiwan Island, the typhoon driven wave nowcasting model in Southeast China Sea is setup. The SWAN (simulating waves nearshore) model is used to calculate wave field and the input wind field is the QSCAT/NCEP (Quick Scatterometer/National Centers for Environmental Prediction) data. The two-dimensional wavelet transform is applied to analyze the X-band radar image of nearshore wave field and it reveals that the observed wave spectrum has shoaling characteristics in frequency domain. The reverse calculation approach of wave spectrum in deep water is proposed and validated with experimental tests. The two-dimensional digital low-pass filter is used to obtain the initialization wave field. Wave data during Typhoon Sinlaku is used to calibrate the data assimilation parameters and test the reverse calculation approach. Data assimilation corrects the significant wave height and the low frequency spectra energy evidently at Beishuang Station along Fujian Province coast, where the entire assimilation indexes are positive in verification moments. The nowcasting wave field shows that the present model can obtain more accurate wave predictions for coastal and ocean engineering in Southeast China Sea.

AN ITERATIVE ALGORITHM FOR OPTIMAL DESIGN OF NON-FREQUENCY-SELECTIVE FIR DIGITAL FILTERS

This paper proposes a novel iterative algorithm for optimal design of non-frequency-selective Finite Impulse Response(FIR) digital filters based on the windowing method.Different from the traditional optimization concept of adjusting the window or the filter order in the windowing design of an FIR digital filter,the key idea of the algorithm is minimizing the approximation error by succes-sively modifying the design result through an iterative procedure under the condition of a fixed window length.In the iterative procedure,the known deviation of the designed frequency response in each iteration from the ideal frequency response is used as a reference for the next iteration.Because the approximation error can be specified variably,the algorithm is applicable for the design of FIR digital filters with different technical requirements in the frequency domain.A design example is employed to illustrate the efficiency of the algorithm.

利用取样功率计确定射频和微波装置的特性

因此,用户可通过改变上述控制对信号在显示器上位置进行控制见表4。这里介绍数字示波器常用的概念。数字示波器都有可调比较电路,

Efficient design method of non-uniform cosine modulated filter bank for digital hearing aids

To promote the performance of the traditional multichannel filter bank which leads to speech quality degradation,an efficient design method of the non-uniform cosine modulated filter bank（CMFB） based on the audiogram for digital hearing aids is proposed. First, a low-pass prototype filter is designed by the linear iterative algorithm. Secondly,the uniform CMFB is achieved on the basis of the principle formulas. Then, the adjacent channels of a uniform filter bank which have low or gradual slopes are merged according to the trend of audiogram of the hearing impaired person. Finally,the corresponding non-uniform CMFB is obtained. Simulation results show that the signal processed by the proposed filter bank is similar to the original signal in a time-domain waveform and spectrogram without significant distortion or difference. The speech quality results show that the personal evaluation of speech quality（PESQ） of non-uniform CMFB is 35% higher than that of the traditional design, and the hearing-aid speech quality index（HASQI） increases by about 40%.

DPF单通道内颗粒物沉积特性的数值模拟

基于离散的格子Boltzmann方法(LBM)耦合格子气元胞(CA)自动机概率模型,结合计算机断层扫描技术与四参数随机生成法获取和生成过滤器薄壁微观数字结构,模拟柴油机颗粒捕集器(DPF)通道内不同入口流速u_(0)下的颗粒物沉积特性.结果表明:不同粒径颗粒群在过滤器壁面的沉积分布很大程度取决于通道气流的u_(0),u_(0)=1 m/s时颗粒群沿着通道壁面沉积分布比较均匀;u_(0)=3 m/s、u_(0)=6 m/s时颗粒更加倾向于沉积在通道的中、后段.颗粒沉积后,过滤器通道的有效宽度减小,与洁净通道相比,轴向速度与压力都有所增加,且u_(0)越大,变化越明显.

DYNAMIC COMPENSATION OF MEASURING SYSTEMS BY SYSTEM IDENTIFICATION APPROACH

A digital filtering method is presented to compensate the dynamic characteristics of measuring systems.The compensation filter has an infinite impulse response property and is designed by system identification approach from the known input output pairs of the measuring system.Applications of this method to eliminating the distortions of measured waveform in transient pulse measurement are investigated.Experimental results show that the measurement errors caused by the sensor are reduced to be very small after the use of the compensation filter.

A Low Power/Area Digital FIR Filter Design Based on PRF Framework

A novel DSP to ASIC (Application Specific Integrated Circuit) architecture design methodology is presented in this paper for reducing power/area consumption. Traditional methods always focus on optimizing hardware structure or algorithm separately. The authors propose a new method called PRF (Paralleling Reducing Folding) framework to combine hardware optimization with algorithm simplification. In the first step, paralleling, unfolding technology is applied to divide one data path into several channels and expose the redundancy of the algorithm. In the second step, reducing, decoupling theory is used to reduce computational complexity. In the last step, folding, time multiplexing method is used to merge similar components. As an exoteric methodology framework, many optimization methods can be integrated into the PRF framework. To optimize a 3N taps FIR (Fincte Impact Response) and obtain a content result, PRF methodology framework is applied.

Four optimal design approaches of high-order finite-impulse response filters based on neural network

Four optimal approaches of high-order finite-impulse response(FIR) digital filters were developed for designing four types filters using neural network algorithms. The solutions were presented as parallel algorithms to approximate the desired frequency response specification. Therefore, these methods avoid matrix inversion, and make a fast calculation of the filter’s coefficients possible. The convergence theorems of these proposed algorithms were presented and proved to illustrate them stable, and the implementation of these methods was described together with some design guidelines. The simulation results show that the ripples of the designed FIR filters are significantly little in the pass-band and stop-band, and the proposed algorithms are of fast convergence.

Normalized Design of Multi-channel LIF Digital Receiver

A design method for parallel processing application on multi-channel low-intermediate-frequency(LIF) digital receiver was presented. It is based on the DSP sub-array with a simple topology and operation timing to evaluate and determine the processing capability and then construct the parallel processing array for multi-channel signals according to the restriction of operation timing. Using this method, the design of multi-channel digital receiver may be simplified. Finally, a design example was used to show how to apply this method.

High-Speed Numeral Filter in Impulse Voltage Measurement

Since the CPU of embed system has some limitation in operating speed, a new filter was put forward which implemented mountain template convolution by performing rectangle template convolution two times. It can obtain time and frequency localization with computational complexity greatly reduced. This algorithm was applied to lightning waveforms (include chopped waveforms) parameter calculation. It simplifies the computation and the results pretreated by this algorithm are in accord with IEC1083-2 completely. It was applied in embed system successfully. Its capability in frequency restraining was researched. The validity of the algorithm was proved in theory when processing lightning waves. The standard sources and the processing results are consistent completely.

Quantitative analysis of the performance of vector tracking algorithms

Vector tracking changes the classical structure of receivers. Combining signal tracking and navigation solution,vector tracking can realize powerful processing capabilities by the fusion technique of receiving channel and feedback correction. In this paper,we try to break through the complicated details of numerical analysis,consider the overall influencing factors of the residual in observed data,and use the intrinsic link between a conventional receiver and a vector receiver. A simple method for performance analysis of the vector tracking algorithm is proposed. Kalman filter has the same steady performance with the classic digital lock loop through the analysis of the relation between gain and band width. The theoretical analysis by the least squares model shows that the reduction of range error is the basis for the superior performance realized by vector tracking. Thus,the bounds of its performance enhancement under weak signal and highly dynamic conditions can be deduced. Simulation results verify the effectiveness of the analysis presented here.

APULSECOMPRESSORIMPLEMENTATIONWITHDSPFORSMALLTBPRODUCTLFMWAVEFORM

This paper describes a pulse compressor implementation with DSP for small Time Bandwidth (TB) product Linear Frequency Modulation (LFM) waveform. It contains the digital generation of the LFM waveform and the digital internally Hamming weighted compression filter. Two methods for suppression of time sidelobe of the digital pulse compressor are employed. First, the LFM waveform is modified by using cubic phase pre distortion for reducing the effect of Fresnel ripples in small TB product LFM waveform. Secondly, anti aliasing filter is used before A/D converter for reducing spectrum skirt level of the returned LFM waveform. The parameters of the compression filter implemented with IMSA100 DSP are programmable. The experiments show that the peak time sidelobe level of the digital pulse compressor is less than -32 dB for TB product of 20.

Digital FIR filter design for bio-signal processing system

Recently,real-time processing systems for bio-signal of the muscles generated by the movement of the user have been developed.Finite impulse response(FIR)filter for bio-signal processing in bio-signal process systems is composed of multiple multiplier and adder of high-area.This makes the chip area increase significantly.To solve this problem,a low-area digital FIR filter is proposed in this paper,which can reduce the chip area.

Digital Filter for Electrocardiogram Preprocessing Based on Microprocessor

This paper proposes a different method to eliminate base wander and power line interference in electrocardiogram, which introduces the integer coefficient filter theory and gives the detail for designing digital filter to remove these two normal noise signals. Signal from the MIT-BIH electrocardiogram database was used to test the performance of the filter. From the test results, the performance of the digital filer is reDT good. The filter coefficient is an integer number, therefore, the filtering algorithm can be successfully implemented on the microprocessor.