scholarly journals VLSI Design and Implementation for Adaptive Filter using LMS Algorithm

Author(s):  
Asit Kumar Subudhi ◽  
Biswajit Mishra ◽  
Mihir N. Mohanty

Adaptive filters, as part of digital signal systems, have been widely used, as well as in applications such as adaptive noise cancellation, adaptive beam forming, channel equalization, and system identification. However, its implementation takes a great deal and becomes a very important field in digital system world. When FPGA (Field Programmable Logic Array) grows in area and provides a lot of facilities to the designers, it becomes an important competitor in the signal processing market. In general FIR structure has been used more successfully than IIR structure in adaptive filters. However, when the adaptive FIR filter was made this required appropriate algorithm to update the filter’s coefficients. The algorithm used to update the filter coefficient is the Least Mean Square (LMS) algorithm which is known for its simplification, low computational complexity, and better performance in different running environments. When compared to other algorithms used for implementing adaptive filters the LMS algorithm is seen to perform very well in terms of the number of iterations required for convergence. This phenomenon can be achieved by a sufficient choice of bit length to represent the filter’s coefficients. This paper presents a lowcost and high performance programmable digital finite impulse response (FIR) filter. It follows the adaptive algorithm used for the development of the system. The architecture employs the computation sharing algorithm to reduce the computation complexity.

Electronics ◽  
2019 ◽  
Vol 8 (5) ◽  
pp. 482
Author(s):  
Mangi Han ◽  
Youngmin Kim

In this study, we implemented a high-performance multichannel repeater, both for FM and T-Digital Multimedia Broadcasting (DMB) signals using a Field Programmable Gate Array (FPGA). In a system for providing services using wireless communication, a radio-shaded area is inevitably generated due to various obstacles. Thus, an electronic device that receives weak or low-level signals and retransmits them at a higher level is crucial. In addition, parallel implementation of digital filters and gain controllers is necessary for a multichannel repeater. When power level is too low or too high, the repeater is required to compensate the power level and ensure a stable signal. However, analog- and software-based repeaters are expensive and they are difficult to install. They also cannot effectively process multichannel in parallel. The proposed system exploits various digital signal-processing algorithms, which include modulation, demodulation, Cascaded Integrator Comb (CIC) filters, Finite Impulse Response (FIR) filters, Interpolated Second Ordered Polynomials (ISOP) filters, and Automatic Gain Controllers (AGCs). The newly proposed AGC is more efficient than others in terms of computation amount and throughput. The designed digital circuit was implemented by using Verilog HDL, and tested using a Xilinx Kintex 7 device. As a result, the proposed repeater can simultaneously handle 40 FM channels and 6 DMB channels in parallel. Output power level is also always maintained by the AGC.


Author(s):  
A Aparna ◽  
T Vigneswaran

This research work proposes the finite impulse response (FIR) filters design using distributed arithmetic architecture optimized for field programmable gate array. To implement computationally efficient, low power, high-speed FIR filter a two-dimensional fully pipelined structure is used. The FIR filter is dynamically reconfigured to realize low pass and high pass filter by changing the filter coefficients. The FIR filter is most fundamental components in digital signal processing for high-speed application. The aim of this research work is to design multiplier-less FIR filter for the requirements of low power and high speed various embedded applications. 


2019 ◽  
Vol 8 (4) ◽  
pp. 11849-11853

FIR (Finite Impulse Response) filters play a significant role in the field of Digital Signal Processing (DSP) to eliminate noise suppression in Electro Cardio Graph (ECG), Imaging devices and the signal stored in analog media. So filter evaluation is accomplished to reduce the noise level. The Filter passes only the desired frequency to pass thereby reducing distortion in the processed signal during measurement. The FIR filter comprises of basic units like adders, multipliers and the delay element for its operations.FIR and IIR are the two types of digital filters chosen based on the range of inputs, complexity and size requirement. Multipliers and adders play a vital role in deterring the performance of FIR filter. In this work, we design and analyze different multiplier and adder for high-performance Fir filter implementation. The Vedic Mathematics is the methods containing 16 Sutras to aid fast mental calculations. In this work, we propose modified Anurupye Vedic multiplier methods with Kogge Stone fast adder for implementation in the direct form FIR filter. This approach provides 1.5% decrease in delay and 10.2% reduced in power, hence increasing speed marginally than previous methods. Along with low power consumption in Very High-Speed Hardware Description Language (VHDL), all the adders and the multiplier topologies are Synthesized using (Xilinx Spartan – 6 FPGA) Trainer Kit and the proposed 8 – Tap FIR filter is executed using this Board


Author(s):  
Noopur Astik

Dynamic partial reconfiguration has evolved as a very prominent state of art for efficient area utilization of <em>Field Programmable Gate Array</em> (FPGA) as well as significant reduction in its overall power consumption when properly used to lessen the idle logic on FPGA. It provides desired results even as the computational complexity increases in the field of Digital Signal Processing. This paper explains Dynamic Partial Reconfiguration (DPR) with an example of Finite Impulse response (FIR) filter of order 10. Initially RTL coding for Direct Form FIR structure is written in Verilog in fixed point format for low pass and high pass filter modules using ISE Design suite. Functioning of the both the modules is verified individually through hardware co-simulation on ZYBO (Zynq Board) from Digilent using Black Box from System Generator. Finally dynamic partial reconfigurable FIR filter with low pass and high pass as reconfigurable modules is implemented on ZYBO using PlanAhead tool. Final comparison of resource utilization with and without DPR is presented


2021 ◽  
Vol ahead-of-print (ahead-of-print) ◽  
Author(s):  
Mohan Kumar ◽  
Ranga Raju

Purpose Digital signal processing (DSP) applications such as finite impulse response (FIR) filter, infinite impulse response and wavelet transformation functions are mainly constructed using multipliers and adders. The performance of any digital applications is dependent on larger size multipliers, area and power dissipation. To optimize power and area, an efficient zero product and feeder register-based multiplier (ZP and FRBM) is proposed. Another challenging task in multipliers is summation of partial products (PP), results in more delay. To address this issue, the modified parallel prefix adder (PPA) is incorporated in multiplier design. In this work, different methods are studied and analyzed for designing FIR filter, optimized with respect to area, power dissipation, speed, throughput, latency and hardware utilization. Design/methodology/approach The distributed arithmetic (DA)-based reconfigurable FIR design is found to be suitable filter for software-defined radio (SDR) applications. The performance of adder and multipliers in DA-FIR filter restricts the area and power dissipation due to their complexity in terms of generation of sum and carry bits. The hardware implementation time of an adder can be reduced by using PPA which is based on Ling equation. The MDA-RFIR filter is designed for higher filter length (N), i.e. N = 64 with 64 taps and this design is developed using Verilog hardware description language (HDL) and implemented on field-programmable gate array. The design is validated for SDR channel equalizer; both RFIR and SDR are integrated as single system and implemented on Artix-7 development board of part name XC7A100tCSG324. Findings The MDA-RFIR for N = 64 is optimized about 33% in terms of area-delay, power-speed product and energy efficiency. The theoretical and practical comparisons have been done, and the practically obtained results are compared with existing DA-RFIR designs in terms of throughput, latency, area-delay, power-speed product and energy efficiency are better about 3.5 times, 31, 45 and 29%, respectively. Originality/value The MDA-RFIR for N = 64 is optimized about 33% in terms of area-delay, power-speed product and energy efficiency.


In DSP the most common function is Finite Impulse Response (FIR) filter which is realized in field Programmable gate Arrays (FPGAs). For efficient Very Large Scale Integration (VLSI) computation systolic FIR filter architecture has attractive models. High speed is the major concern for fast computation in real time Digital Signal Processing (DSP) applications. In conventional systolic FIR filter method uses general array multiplier structure which takes more time to compute the process with high design complexity with less power. To overcome this problem the systolic FIR filter utilizing Bypass Feed Direct Multiplier(BFDM) is proposed. The proposed method 16 tap systolic FIR parallel processing offers less delay with less design complexity which is used in image and signal processing applications. The proposed method is simulated using Xilinx ISE 12.4 ISE tool and the functions are evaluated by MODELSIM 6.3C.


Electronics ◽  
2021 ◽  
Vol 10 (8) ◽  
pp. 884
Author(s):  
Stefano Rossi ◽  
Enrico Boni

Methods of increasing complexity are currently being proposed for ultrasound (US) echographic signal processing. Graphics Processing Unit (GPU) resources allowing massive exploitation of parallel computing are ideal candidates for these tasks. Many high-performance US instruments, including open scanners like ULA-OP 256, have an architecture based only on Field-Programmable Gate Arrays (FPGAs) and/or Digital Signal Processors (DSPs). This paper proposes the implementation of the embedded NVIDIA Jetson Xavier AGX module on board ULA-OP 256. The system architecture was revised to allow the introduction of a new Peripheral Component Interconnect Express (PCIe) communication channel, while maintaining backward compatibility with all other embedded computing resources already on board. Moreover, the Input/Output (I/O) peripherals of the module make the ultrasound system independent, freeing the user from the need to use an external controlling PC.


2021 ◽  
Vol 34 (1) ◽  
pp. 133-140
Author(s):  
Teimour Tajdari

This study investigates the ability of recursive least squares (RLS) and least mean square (LMS) adaptive filtering algorithms to predict and quickly track unknown systems. Tracking unknown system behavior is important if there are other parallel systems that must follow exactly the same behavior at the same time. The adaptive algorithm can correct the filter coefficients according to changes in unknown system parameters to minimize errors between the filter output and the system output for the same input signal. The RLS and LMS algorithms were designed and then examined separately, giving them a similar input signal that was given to the unknown system. The difference between the system output signal and the adaptive filter output signal showed the performance of each filter when identifying an unknown system. The two adaptive filters were able to track the behavior of the system, but each showed certain advantages over the other. The RLS algorithm had the advantage of faster convergence and fewer steady-state errors than the LMS algorithm, but the LMS algorithm had the advantage of less computational complexity.


Author(s):  
M. Yasin ◽  
Pervez Akhtar

Purpose – The purpose of this paper is to analyze the convergence performance of Bessel beamformer, based on the design steps of least mean square (LMS) algorithm, can be named as Bessel LMS (BLMS) algorithm. Its performance is compared in adaptive environment with LMS in terms of two important performance parameters, namely; convergence and mean square error. The proposed BLMS algorithm is implemented on digital signal processor along with antenna array to make it smart in wireless sensor networks. Design/methodology/approach – Convergence analysis is theoretically developed and verified through MatLab Software. Findings – Theoretical model is verified through simulation and its results are shown in the provided table. Originality/value – The theoretical model can obtain validation from well-known result of Wiener filter theory through principle of orthogonality.


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