Finite Impulse Response Digital Filters Research Articles

Research on Ultra-wide Bandwidth Low-frequency Signal Channelization for Xinjiang 110 m Radio Telescope

Aiming at the subband division of ultra-wide bandwidth low-frequency (UWL) signal (frequency coverage range: 704–4032 MHz) of the Xinjiang 110 m QiTai radio Telescope (QTT), a scheme of ultra-wide bandwidth signal is designed. First, we analyze the effect of different window functions such as the Hanning window, Hamming window, and Kaiser window on the performance of finite impulse response (FIR) digital filters, and implement a critical sampling polyphase filter bank (CS-PFB) based on the Hamming window FIR digital filter. Second, we generate 3328 MHz simulation data of ultra-wideband pulsar baseband in the frequency range of 704–4032 MHz using the ultra-wide bandwidth pulsar baseband data generation algorithm based on the 400 MHz bandwidth pulsar baseband data obtained from Parkes CASPSR observations. Third, we obtain 26 subbands of 128 MHz based on CS-PFB and the simulation data, and the pulse profile of each subband by coherent dispersion, integration, and folding. Finally, the phase of each subband pulse profile is aligned by non-coherent dedispersion, and to generate a broadband pulse profile, which is basically the same as the pulse profile obtained from the original data using DSPSR. The experimental results show that the scheme for the QTT UWL receiving system is feasible, and the proposed channel algorithm in this paper is effective.

A spectrum-of-spectrum filtering method to extract direct and multipath arrivals from simulations and measurements

In measurements and numerical modelling of wave propagation, undesired interference between direct and multipath arrivals can be reduced using Fourier-based signal processing methods. Existing methods, such as cepstral analysis and time-signal gating, are not applicable to all cases. Here, an alternative Fourier-based signal processing method is presented, called spectrum-of-spectrum (SoS) filtering. Its main advantage over existing methods is its ability to extract single direct or multipath arrivals for relatively short propagation distances even when subsequent arrivals do not become successively weaker. The method is based on the following steps:•Apply a lowpass filter to the real and imaginary parts of an input frequency spectrum individually, using a digital finite impulse response (FIR) filter in the frequency domain.•Recombine the filtered real and imaginary parts of the frequency spectrum to get the frequency spectrum of the direct arrival.•For extraction of the first multipath arrival, subtract the filtered frequency spectrum from the input frequency spectrum and repeat the previous steps. Repeat multiple times to extract subsequent multipath arrivals.

Effects of Sampled-Values Data Quality on Responses of Time-Frequency Based Digital Relays

Time-frequency based digital protective relays can offer several advantages of high accuracy, improved reliability, and fast response over other protective relays. These protective relays employ half-band digital finite impulse response (FIR) filters to process currents and/or voltages to extract fault signatures, which facilitate accurate detection and identification of fault events. Coefficients of the employed half-band digital FIR filters are determined by wavelet or phaselet functions. The signal processing part in a time-frequency based digital protective relay processes samples of measured currents and/or voltages. As a result, attention has to be paid to the measurement, digitization, and communication of current and voltage data. Power systems utilize the IEC 61850 standard to implement data communication between measurement, control, and protection devices. This paper assesses the impacts of the IEC 61850 sampled values on the accuracy and response speed of time-frequency based digital protective relays. The presented assessment is conducted for two sample cases; the transformer differential protection and arcing current fault detection and identification. Test results show that the accuracy and response speed of time-frequency based digital protective relays are maintained, when measured currents and voltages are communicated using the IEC 61850 sampled values.

AlgDiff: an open source toolbox for the design, analysis and discretisation of algebraic differentiators

Abstract Algebraic differentiators have attracted much interest in recent years. Their simple implementation as classical finite impulse response digital filters and systematic tuning guidelines may help to solve challenging problems, including, but not limited to, nonlinear feedback control, model-free control, and fault diagnosis. This contribution introduces the open source toolbox AlgDiff for the design, analysis and discretisation of algebraic differentiators.

Research on Channelization Techniques of Radio Astronomical Wideband Signal with Oversampled Polyphase Filter Banks

Digital channelization decomposes a wideband signal into multiple adjacent sub-bands using Parallel Technology. Channelization can effectively reduce the pressure on the radio astronomy digital backends system and make wideband signal processing possible. Aiming at the problems of signal attenuation at sub-band edge, spectral leakage and aliasing encountered in wideband signal channelization, algorithms to reduce the problems are studied. We design a Critically Sampled Polyphase Filter Bank (CS-PFB) based on the Finite Impulse Response digital filter with a Hamming Window and systematically analyze the frequency response characteristics of the CS-PFB. Based on the channelized structure of the CS-PFB, an OverSampled Polyphase Filter Bank (OS-PFB) is designed by data reuse, and the filtering frequency response characteristics of CS-PFB and OS-PFB are compared and analyzed. Using the wideband baseband data generated by the CASPSR (Collaboration for Astronomy Signal processing and electronics research Parkes Swinburne Recorder), we implement sub-band division and 16-band output of these data based on the 2× oversampling OS-PFB, and the problem of spectrum inversion in the sub-bands is corrected. After removing 25% of redundant data in the head and tail of each sub-band, we recombine the sub-bands into a wideband. The wideband signal is almost identical to the original observed signal. Therefore, the experimental results show that the OS-PFB can improve the channel response. For the 400 MHz baseband data of J0437-4715, we compare the pulse profile obtained from the original baseband data with the pulse profile obtained after the channelization and recombination. The phase and amplitude information of the pulse profiles are consistent, which verifies the correctness of our channelization algorithm.

Hybrid Protection of Digital FIR Filters

A digital finite impulse response (FIR) filter is a ubiquitous block in digital signal processing applications and its behavior is determined by its coefficients. To protect filter coefficients from an adversary, efficient obfuscation techniques have been proposed, either by hiding them behind decoys or replacing them by key bits. In this article, we initially introduce a query attack that can discover the secret key of such obfuscated FIR filters, which could not be broken by the existing prominent attacks. Then, we propose a first of its kind hybrid technique, including both hardware obfuscation and logic locking using a point function for the protection of parallel direct and transposed forms of digital FIR filters. Experimental results show that the hybrid protection technique can lead to FIR filters with higher security while maintaining the hardware complexity competitive or superior to those locked by prominent logic locking methods. It is also shown that the protected multiplier blocks and FIR filters are resilient to existing attacks. The results on different forms and realizations of FIR filters show that the parallel direct form FIR filter has a promising potential for a secure design.

Design and Implementation of Communication Digital FIR Filter for Audio Signals on the FPGA Platform

Digital Signal Processing (DSP) in telecommunications is widely used for its outstanding performance. Hence, optimal results would be obtained with the sophisticated design and implementation of the digital Finite Impulse Response (FIR) filter. In this paper, the design and implementation of the FIR filter for processing the audio signal are obtained through the Field-Programmable Gate Array (FPGA) platform -Altera DE1 board. The design is divided into three primary blocks, which are the S2P Adapter block, Codec initialization block, as well as FIR filter block, which were then interfaced together; then, every block was compiled and simulated in order to obtain accurate results. The whole system was built to be functional, and finally, the frequency response was obtained. 100% significant accuracy and high quality are examined for implementing the filter in FPGA platform chip board programmed in (VHDL).

Synthesis of the integer FIR filters with short coefficient word length

The integer simulation and development finite impulse response (FIR) filters taking into account the possibilities of their realization on digital integer platforms are considered. The problem statement and solution of multifunctional synthesis of digital FIR filters such a problem on the basis of the numerical methods of integer nonlinear mathematical programming are given. As an several examples, the problem solution of synthesis FIR-filters with short coefficient word length has been given. The analysis of their characteristics is resulted. The paper discusses issues of modeling and synthesis of digital FIR filters with provision for the possibilities of their implementation on digital platforms with integer computation arithmetic. The formulation of the problem of multifunctional synthesis of cascade FIR filters using the methods of integer nonlinear mathematical programming is given. The efficiency of this approach is illustrated by examples of solving the problems of synthesizing integer FIR filters with a minimum coefficient word length. The analysis of the synthesized filter characteristics is made.

Optimization Approach in Window Function Design for Real-Time Filter Applications

Eliminating the Gibbs oscillations that occur during the Finite Impulse Response (FIR) digital filter design with the Fourier Series method will ensure correct filtering. For this reason, the development of the window improves the performance of the filter and, therefore, the system. In this study, the cosh window function is designed using Particle Swarm Optimization, which is a preferred optimization method in many areas. Thus, alternatives to the standard results obtained from the existing traditional calculations will be produced, and different windows that perform the same function will be obtained. In addition, exponential and cosh window functions were designed in LabVIEW environment, which is a graphical programming language-based program, and the designed windows were analyzed at different parameter values. LabVIEW provides a fast and easy programming environment, and it provides the opportunity to realize real-time applications with its external hardware. Utilizing this feature, the amplitude spectrum of cosh window designed in LabVIEW is displayed in real time for different window parameter values. As a result, FIR digital filters were designed using cosh window based on optimization and the cosh window designed in LabVIEW, and the distorted EEG signal was filtered using these filters and displayed in real time.

FIR digital filter design based on improved artificial bee colony algorithm

The traditional swarm intelligence optimization algorithm is prone to fall into local optimal solutions in finite impulse response (FIR) digital filter design and has slow convergence speed. In order to optimize the design of FIR filter, a FIR digital filter design method based on improved bee colony (ABC) algorithm is proposed. This improved ABC algorithm can adaptively adjust the step size of the selected neighborhood of nectar source location. At the same time, the information of the global optimal solution is used to guide the search of candidate solution, which improves the global search ability of the algorithm. The improved ABC algorithm can balance the conflict between local search ability and global search ability, so it can achieve better optimization effect. The time and space complexity of the algorithm is analyzed in detail. Then, the improved ABC algorithm is used to design three typical FIR digital filters, namely low-pass, band-pass and band-stop filter. The performance of the designed filter is tested by simulation experiments. The experimental results show that compared with other state-of-the-art optimization algorithms, the proposed FIR filter design method has achieved better effect and performance. Meanwhile, the proposed design method has shorter optimization time. The superiority of the proposed method is verified.

FIR Filter Design using Raised Semi-ellipse Window Function

In this paper, a new two-parameter window function - Raised semi-ellipse (RSE) is proposed. The window is obtained from a fixed elliptical window known as Semi-ellipse window by raising the radius of the minor axis by the parameter (β), and applied for the design of finite impulse response (FIR) digital filters. The spectral parameters of the proposed window are determined first and compared with the Kaiser window – a 2-parameter adjustable window. Subsequently, in its application in filter design with an established design algorithm, the newly proposed adjustable window is compared to the Semi-ellipse window to examine its improvement and also the Kaiser window to compare its performance with a commonly used adjustable window. The filter simulation results show that the filters designed with the proposed window can provide more reduced ripples than the Kaiser window for prescribed spectral characteristics.

A Low Complexity and High Modularity Design for Continuously Variable Bandwidth Digital Filters

ABSTRACT Digital filters with variable bandwidth can be used for a variety of applications. Arbitrary change in the bandwidth of a digital Finite Impulse Response (FIR) filter can be acquired using sampling rate converters. In this paper, a sampling rate converter is proposed which is generated from Pascal structure, a fractional delay filter having low hardware complexity and high modularity. The proposed sampling rate converter requires lesser number of multipliers for implementation when compared with the sampling rate converters in the literature. A low pass filter having a single bandwidth sandwiched between two sampling rate converters can contribute multiple bandwidths in such a way that each bandwidth is an arbitrary variation of the original bandwidth. A two stage Frequency response masking approach (FRM) is used for the hardware efficient design of the original low pass filter. A low complexity and high modularity novel design for a continuously varying bandwidth of a digital FIR filter is proposed in this paper using the proposed sampling rate converter. The modularity of the Pascal structure can be used to control both pass band ripple as well as stop band attenuation of the continuously variable bandwidth FIR filter design. Different communication standards in a Software defined radio (SDR) channelizer is realized using the proposed design of continuously variable bandwidth filter.

Design and implementation of two-dimensional digital finite impulse response filter using very high speed integrated circuit hardware description language

<span>The main purpose of this paper is to design a two-dimensional digital finite impulse response (FIR) filter using data broadcast and non-broadcast structure. The implementation of two-dimensional digital FIR filter is done using very high speed integrated circuit hardware description language (VHDL). Rectangular window method is used for calculating 2D digital FIR filter coefficients for data broadcast and non-broadcast structure. The coefficients of the one-dimensional digital FIR filter are obtained using the MATLAB filter design and analysis (FDA) tool for two different cut-off frequencies and are multiplied to get the necessary coefficient for the two-dimensional FIR filter to be designed; the simulation is done on Artix-7 series field programmable gate array (FPGA), target device (xc7a35t-cpg236) using Vivadov.2015.2. The proposed design reduces the area utilization and the power consumption when compared with the existing literature. The experimental result shows that the power consumption is improved by 97% and there is an improvement of 24% in area utilization for the two-dimensional with and without data broadcast one dimensional FIR filter structures.</span>

Noise Shaping for Phased Array Beamforming

Quantization of phase and/or amplitude has far-reaching effects on the radiation characteristics of the phased array (PA), including gain, minor lobe level, and point deviation. Traditionally, one common method to address such a nonlinear distortion is using random-phasing to interrupt the error periodicity. Here, we show that the distortion due to the quantization can be better remediated by spectrally shaping the error compared to the random-phasing (dithering) approaches. We adapted the method for phase-only and amplitude-phase synthesis of planar array designed based on analog beamforming (ABF). To do that, for the first time, 2-D real- and complex-coefficient minimum-phase digital finite impulse response (FIR) filters are designed based on the discrete Hilbert transform (DHT) method. In particular, the digital filter design for phase-only synthesis is comprehensively investigated, respecting the error spectra in the beamspace domain. It is shown that by pushing the error out of the so-called visible region, the decrease of antenna directivity due to the quantization can be compensated to some extent, which provides a quite advantage over the uniform distribution of error. For some cases, pushing the error out of the visible region might be impossible. For such cases, we proposed using the spaced-notches filter. It is also shown that the method is on maximum efficacy when both phase and amplitude of the excitation signal are controllable. Thus, complex-valued noise shaping (CV-NS) can be exploited for the phase-amplitude synthesis of the PA, showing quite promising performance.

FPGA implementation of high performance digital FIR filter design using a hybrid adder and multiplier

ABSTRACT The energetic growth in portable multimedia and mobile communication system has increased the requirement of high-speed signal processing system with compact area and power consumption. Finite impulse response(FIR) filters are broadly used in image, signal, speech and video signal processing, medical electronics, noise filtering, mobile communication and many other fields. The performance of the whole signal processing system with FIR filter depends on the basic building block of multiplier and adders. Hence, the hybrid FIR filter is proposed to improve the speed of the signal processing system using hybrid adder and hybrid multiplier. In this technique, the basic hybrid adder is designed with the help of 2-bit adders, BEC and 4:1 Multiplexer. Also, the hybrid multiplier is designed based on partial products of two consecutive multiplicand bits which are added at same time using Han-Carlson, Weinberger and Ling adder. The proposed FIR filter is functionally verified and synthesised using Xilinx ISE simulator and is implemented in Spartan 6 FPGA boards. The result shows that the delay of FIR filter using the proposed multiplier is improved by 26.51%, 15.59%, 15.83% and 2.79% as compared with CLA, conventional CSA, CSA-BK-BEC and the proposed adder with array multiplier, respectively.

Design and Performance Analysis of Improved FIR Filter using Ultra-Scale FPGA

It is discussed in many studies and demonstrated in many pieces of research that based on certain applications, analog design of filter has several issues including complex design, re-use limitations, and accuracy of generating the output at various frequencies. Therefore, instead of analog filter design, the digital design of the filter is preferred for both Finite and Infinite Impulse Response Filter. This paper demonstrates the design of the digital Finite Impulse Response (FIR) filter designed is demonstrated using Ultra-Scale Field Programming Gate Array (FPGA) having chip XCKU3P. The filter is designed using a coefficient multiplier via Canonic Signed Digit (CSD) Technique. The optimized design of the digital filter is conducted via real-time implementation is performed using Ultra-Scale FPGA. The filter is designed and evaluated with an ordinary filter at 10 MHz and 10 GHz frequencies. The performance analysis of the system is illustrated using the response rate at the bitstream of 16-bit. In the results, it is demonstrated that for 10 MHz frequency design FIR filter in FPGA the 30% faster response filter is achieved at for 10 GHz, the 15% faster response is achieved at the I/O standard of Low Voltage Complementary Metal Oxide Semiconductor (LVCOMS). The optimization of 30% in terms of the response time of the filter is attained using the proposed work. The proposed improved FIR filter design using Ultra-Scale FPGA helps in increasing design performance to increase the speed of overall response of FIR filter that is lacking in ordinary Filters.

Low-Complexity Chromatic Dispersion Equalization FIR Digital Filter for Coherent Receiver

This paper proposes a novel and efficient low-complexity chromatic dispersion equalizer (CDE) based on finite impulse response (FIR) filter architecture for polarization-multiplexed coherent optical communication systems. The FIR filter coefficients are optimized by weights to reduce the energy leakage caused by the truncation effect, and then quantization is used uniformly to reduce the number of real number additions and real number multiplications by utilizing the diversity of the quantized coefficients. Using Optisystem 15 to build a coherent optical communication system for simulation and experimental demonstration, the results show that after the filter coefficients are optimized by weights. Compared with the time-domain chromatic dispersion equalizer (TD-CDE), the proposed design has a lower bit error rate (BER) and better equalization effect. When the transmission distance is 4000 km and the system quantization stages M = 16, the multiplication operation and addition operations reduce computing resources by 99% and 43%, and the BER only increases by 5%. Compared with frequency-domain chromatic dispersion equalizer (FD-CDE), widely used in long-distance communication, the multiplication operation reduces computing resources by 30%. The proposed method provides a new idea for high-performance CDE in long-distance coherent optical communication systems.

FPGA implantations of TRNG architecture using ADPLL based on FIR filter as a loop filter

This article describes about the design, implementation, and analysis of a true random number generator (TRNG) employing an all-digital phase-locked loop (ADPLL) based on a finite impulse response (FIR) filter as the digital loop filter and implemented on the Artix 7(XC7A35T-CPG236-1) field programmable gate array (FPGA) board using the Xilinx Vivado v.2015.2 design suite. The coefficients of a 3rd-order broadcast low-pass digital FIR filter are computed using the Keiser window method. The MATLAB-FDA tool is used to calculate the filter coefficient. After reducing the bias in the sequence using the XOR-corrector post-processing approach, the suggested ADPLL-based TRNG designs created an unbiased stochastic random number with an overall throughput of 200 Mbps for both designs. The first proposed FIR-ADPLL-based TRNG design (FAT-1) consumes 0.072 W of power, whereas the second proposed FIR-ADPLL-based TRNG design (FAT-2) consumes 0.074 W. The resulting bitstream is verified for randomness using national institute of standards and technology (NIST) test following post-processing. Digital storage oscilloscope (DSO) is interfaced to the Artrix-7 FPGA board in order to capture the TRNG output waveforms. Both the proposed FIR-based ADPLL-TRNG designs passed the NIST SP 800-22 test, indicating that they are well-suited for a variety of industrial applications, including network security, cybersecurity, banking security, smart cards, radio-frequency identification tags, Internet of Things, and Industrial Internet of Things.

Knowledge-driven feature component interpretable network for motor imagery classification

Objective. The end-to-end convolutional neural network (CNN) has achieved great success in motor imagery (MI) classification without a manual feature design. However, all the existing deep network solutions are purely datadriven and lack interpretability, which makes it impossible to discover insightful knowledge from the learned features, not to mention to design specific network structures. The heavy computational cost of CNN also makes it challenging for real-time application along with high classification performance. Approach. To address these problems, a novel knowledge-driven feature component interpretable network (KFCNet) is proposed, which combines spatial and temporal convolution in analogy to independent component analysis and a power spectrum pipeline. Prior frequency band knowledge of sensory-motor rhythms has been formulated as band-pass linear-phase digital finite impulse response filters to initialize the temporal convolution kernels to enable the knowledge-driven mechanism. To avoid signal distortion and achieve a linear phase and unimodality of filters, a symmetry loss is proposed, which is used in combination with the cross-entropy classification loss for training. Besides the general prior knowledge, subject-specific time-frequency property of event-related desynchronization and synchronization has been employed to construct and initialize the network with significantly fewer parameters. Main results. Comparison of experiments on two public datasets has been performed. Interpretable feature components could be observed in the trained model. The physically meaningful observation could efficiently assist the design of the network structure. Excellent classification performance on MI has been obtained. Significance. The performance of KFCNet is comparable to the state-of-the-art methods but with much fewer parameters and makes real-time applications possible.

Flexible Sub-bands F-OFDM Configured for Spectrum Efficiency Enhancement in 5G System

The new product of wireless communication systems, the Fifth generation (5G), promises higher data rates and Spectrum Efficiency (SE) enhancements to support the communication of heterogeneous services. The Filtered-OFDM (F-OFDM) technique was proposed as the strongest candidate waveform for the physical layer in 5G to fulfill these requirements. In F-OFDM, the whole band is split into narrow sub-bands, each filtered by a digital Finite Impulse Response (FIR) filter with different specifications to increase the spectrum utilization and allow for asynchronous transmission. This paper proposes a novel F-OFDM design waveform for the first time with four sub-bands in equal and unequal sub-band sizes of seven kinds of window sinc filters and a variety of numerology designs to observe SE enhancement using Matlab-Simulink Software. Simulation results show that F-OFDM can reduce Out-Of-Band Emission (OOBE) and achieve SE of about (5%-6%) higher than conventional OFDM for equal and unequal sized sub-bands, respectively, by optimizing the guard band between the designed sub-bands, which achieves 5G guard band requirements.