This paper aims at implementation of a low power adaptive FIR filter based on distributed arithmetic (DA) with
low power, high throughput, and low area. Least Mean Square (LMS) Algorithm is used to update the weight
and decrease the mean square error between the current filter output and the desired response. The pipelined
Distributed Arithmetic table reduces switching activity and hence it reduces power. The power consumption is
reduced by keeping bit-clock used in carry-save accumulation much faster than clock of rest of the operations.
We have implemented it in Quartus II and found that there is a reduction in the total power and the core dynamic
power by 31.31% and 100.24% respectively when compared with the architecture without DA table
vFORTRAN is used as a numerical and scientific computing language. The main objective of the lab work is to understand FORTRAN language using which we solve simple numerical problems and compare different methodologies. Through this project we make use of various functions of FORTRAN and solve a simple projectile problem and also LAPACK library to solve a tridiagonal matrix problem. We use DGESV and DGTSV functions to make it possible. The given problems are built and compiled using a free integrated development environment called CODE::BLOCKS [1] which is an open source platform for FORTRAN and C.
A landing gear assembly consists of various components viz. Lower side stay, Upperside stay, Locking actuators, Extension actuators, Tyres, and Locking pins to name a few. Each unit having a specific operation to deal with, in this project the main unit being studied is the lower brace. The primary objective is to analyse stresses in the element of the lower brace unit using strength of materials or RDM method and Finite Element Method (FEM) and compare both. Using the obtained data a suitable material is proposed for the component. The approach used here is to study the overall behaviour of the element by taking up each aspect, finally summing up the total effect of all the aspects in the functioning of the element.
Discrete-wavelet-transform recursive inverse algorithm using second-order est...TELKOMNIKA JOURNAL
The recursive-least-squares (RLS) algorithm was introduced as an alternative to LMS algorithm with enhanced performance. Computational complexity and instability in updating the autocolleltion matrix are some of the drawbacks of the RLS algorithm that were among the reasons for the intrduction of the second-order recursive inverse (RI) adaptive algorithm. The 2nd order RI adaptive algorithm suffered from low convergence rate in certain scenarios that required a relatively small initial step-size. In this paper, we propose a newsecond-order RI algorithm that projects the input signal to a new domain namely discrete-wavelet-transform (DWT) as pre step before performing the algorithm. This transformation overcomes the low convergence rate of the second-order RI algorithm by reducing the self-correlation of the input signal in the mentioned scenatios. Expeirments are conducted using the noise cancellation setting. The performance of the proposed algorithm is compared to those of the RI, original second-order RI and RLS algorithms in different Gaussian and impulsive noise environments. Simulations demonstrate the superiority of the proposed algorithm in terms of convergence rate comparedto those algorithms.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
I am Felix T. I am a Digital Signal Processing Assignment Expert at matlabassignmentexperts.com. I hold a Ph.D. in Matlab, University of Greenwich, UK. I have been helping students with their homework for the past 4 years. I solve assignments related to Digital Signal Processing.
Visit matlabassignmentexperts.com or email info@matlabassignmentexperts.com.
You can also call on +1 678 648 4277 for any assistance with Digital Signal Processing Assignments.
I am Martin J. I am a Digital Signal Processing Assignment Expert at matlabassignmentexperts.com. I hold a Ph.D. in Matlab, Arizona University, USA. I have been helping students with their homework for the past 6 years. I solve assignments related to Digital Signal Processing.
Visit matlabassignmentexperts.com or email info@matlabassignmentexperts.com.
You can also call on +1 678 648 4277 for any assistance with Digital Signal Processing Assignments.
FINITE DIFFERENCE MODELLING FOR HEAT TRANSFER PROBLEMSroymeister007
This report provides a practical overview of numerical solutions to the heat equation using the finite difference method (FDM). The forward time, centered space (FTCS), the backward time, centered space (BTCS), and Crank-Nicolson schemes are developed, and applied to a simple problem in1volving the one-dimensional heat equation. Complete, working Matlab and FORTRAN codes for each program are presented. The results of running the codes on finer (one-dimensional) meshes, and with smaller time steps are demonstrated. These sample calculations show that the schemes realize theoretical predictions of how their truncation errors depend on mesh spacing and time step. The Matlab codes are straightforward and allow us to see the differences in implementation between explicit method (FTCS) and implicit methods (BTCS). The codes also allow us to experiment with the stability limit of the FTCS scheme.
vFORTRAN is used as a numerical and scientific computing language. The main objective of the lab work is to understand FORTRAN language using which we solve simple numerical problems and compare different methodologies. Through this project we make use of various functions of FORTRAN and solve a simple projectile problem and also LAPACK library to solve a tridiagonal matrix problem. We use DGESV and DGTSV functions to make it possible. The given problems are built and compiled using a free integrated development environment called CODE::BLOCKS [1] which is an open source platform for FORTRAN and C.
A landing gear assembly consists of various components viz. Lower side stay, Upperside stay, Locking actuators, Extension actuators, Tyres, and Locking pins to name a few. Each unit having a specific operation to deal with, in this project the main unit being studied is the lower brace. The primary objective is to analyse stresses in the element of the lower brace unit using strength of materials or RDM method and Finite Element Method (FEM) and compare both. Using the obtained data a suitable material is proposed for the component. The approach used here is to study the overall behaviour of the element by taking up each aspect, finally summing up the total effect of all the aspects in the functioning of the element.
Discrete-wavelet-transform recursive inverse algorithm using second-order est...TELKOMNIKA JOURNAL
The recursive-least-squares (RLS) algorithm was introduced as an alternative to LMS algorithm with enhanced performance. Computational complexity and instability in updating the autocolleltion matrix are some of the drawbacks of the RLS algorithm that were among the reasons for the intrduction of the second-order recursive inverse (RI) adaptive algorithm. The 2nd order RI adaptive algorithm suffered from low convergence rate in certain scenarios that required a relatively small initial step-size. In this paper, we propose a newsecond-order RI algorithm that projects the input signal to a new domain namely discrete-wavelet-transform (DWT) as pre step before performing the algorithm. This transformation overcomes the low convergence rate of the second-order RI algorithm by reducing the self-correlation of the input signal in the mentioned scenatios. Expeirments are conducted using the noise cancellation setting. The performance of the proposed algorithm is compared to those of the RI, original second-order RI and RLS algorithms in different Gaussian and impulsive noise environments. Simulations demonstrate the superiority of the proposed algorithm in terms of convergence rate comparedto those algorithms.
IJERA (International journal of Engineering Research and Applications) is International online, ... peer reviewed journal. For more detail or submit your article, please visit www.ijera.com
I am Felix T. I am a Digital Signal Processing Assignment Expert at matlabassignmentexperts.com. I hold a Ph.D. in Matlab, University of Greenwich, UK. I have been helping students with their homework for the past 4 years. I solve assignments related to Digital Signal Processing.
Visit matlabassignmentexperts.com or email info@matlabassignmentexperts.com.
You can also call on +1 678 648 4277 for any assistance with Digital Signal Processing Assignments.
I am Martin J. I am a Digital Signal Processing Assignment Expert at matlabassignmentexperts.com. I hold a Ph.D. in Matlab, Arizona University, USA. I have been helping students with their homework for the past 6 years. I solve assignments related to Digital Signal Processing.
Visit matlabassignmentexperts.com or email info@matlabassignmentexperts.com.
You can also call on +1 678 648 4277 for any assistance with Digital Signal Processing Assignments.
FINITE DIFFERENCE MODELLING FOR HEAT TRANSFER PROBLEMSroymeister007
This report provides a practical overview of numerical solutions to the heat equation using the finite difference method (FDM). The forward time, centered space (FTCS), the backward time, centered space (BTCS), and Crank-Nicolson schemes are developed, and applied to a simple problem in1volving the one-dimensional heat equation. Complete, working Matlab and FORTRAN codes for each program are presented. The results of running the codes on finer (one-dimensional) meshes, and with smaller time steps are demonstrated. These sample calculations show that the schemes realize theoretical predictions of how their truncation errors depend on mesh spacing and time step. The Matlab codes are straightforward and allow us to see the differences in implementation between explicit method (FTCS) and implicit methods (BTCS). The codes also allow us to experiment with the stability limit of the FTCS scheme.
Area efficient parallel LFSR for cyclic redundancy check IJECEIAES
Cyclic Redundancy Check (CRC), code for error detection finds many applications in the field of digital communication, data storage, control system and data compression. CRC encoding operation is carried out by using a Linear Feedback Shift Register (LFSR). Serial implementation of CRC requires more clock cycles which is equal to data message length plus generator polynomial degree but in parallel implementation of CRC one clock cycle is required if a whole data message is applied at a time. In previous work related to parallel LFSR, hardware complexity of the architecture reduced using a technique named state space transformation. This paper presents detailed explaination of search algorithm implementation and technique to find number of XOR gates required for different CRC algorithms. This paper presents a searching algorithm and new technique to find the number of XOR gates required for different CRC algorithms. The comparison between proposed and previous architectures shows that the number of XOR gates are reduced for CRC algorithms which improve the hardware efficiency. Searching algorithm and all the matrix computations have been performed using MATLAB simulations.
Engineering Research Publication
Best International Journals, High Impact Journals,
International Journal of Engineering & Technical Research
ISSN : 2321-0869 (O) 2454-4698 (P)
www.erpublication.org
On selection of periodic kernels parameters in time series predictioncsandit
In the paper the analysis of the periodic kernels parameters is described. Periodic kernels can
be used for the prediction task, performed as the typical regression problem. On the basis of the
Periodic Kernel Estimator (PerKE) the prediction of real time series is performed. As periodic
kernels require the setting of their parameters it is necessary to analyse their influence on the
prediction quality. This paper describes an easy methodology of finding values of parameters of
periodic kernels. It is based on grid search. Two different error measures are taken into
consideration as the prediction qualities but lead to comparable results. The methodology was
tested on benchmark and real datasets and proved to give satisfactory results.
K-Sort: A New Sorting Algorithm that Beats Heap Sort for n 70 Lakhs!idescitation
Sundararajan and Chakraborty [10] introduced a new
version of Quick sort removing the interchanges. Khreisat
[6] found this algorithm to be competing well with some other
versions of Quick sort. However, it uses an auxiliary array
thereby increasing the space complexity. Here, we provide a
second version of our new sort where we have removed the
auxiliary array. This second improved version of the algorithm,
which we call K-sort, is found to sort elements faster than
Heap sort for an appreciably large array size (n 70,00,000)
for uniform U[0, 1] inputs.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
FORTRAN is used as a numerical and scientific computing language. The main objective of the lab work is to understand FORTRAN language using which we solve simple numerical problems and compare different methodologies. Through this project we make use of various functions of FORTRAN and solve a FDM simple heat equation problem applying different conditions viz. Dirichlet and Von Neumann. The given problems are solved analytically then built and compiled using a free integrated development environment called CODE::BLOCKS [1] which is an open source platform for FORTRAN and C.
Using several mathematical examples from three different authors in texts from different courses this paper illustrates the easier way to avoid confusions and always get the correct results with the least effort was to use the proposed Excel Gamma function explained in detail for the proper use of the Q(z) and ercf(x) functions in most communication courses. The paper serves as a tutorial and introduction for such functions
A NEW PARALLEL ALGORITHM FOR COMPUTING MINIMUM SPANNING TREEijscmc
Computing the minimum spanning tree of the graph is one of the fundamental computational problems. In
this paper, we present a new parallel algorithm for computing the minimum spanning tree of an undirected
weighted graph with n vertices and m edges. This algorithm uses the cluster techniques to reduce the
number of processors by fraction 1/f (n) and the parallel work by the fraction O ( 1 lo g ( f ( n )) ),where f (n) is an
arbitrary function. In the case f (n) =1, the algorithm runs in logarithmic-time and use super linear work on
EREWPRAM model. In general, the proposed algorithm is the simplest one.
HEATED WIND PARTICLE’S BEHAVIOURAL STUDY BY THE CONTINUOUS WAVELET TRANSFORM ...cscpconf
Nowadays Continuous Wavelet Transform (CWT) as well as Fractal analysis is generally used for the Signal and Image processing application purpose. Our current work extends the field of application in case of CWT as well as Fractal analysis by applying it in case of the agitated wind particle’s behavioral study. In this current work in case of the agitated wind particle, we have mathematically showed that the wind particle’s movement exhibits the “Uncorrelated” characteristics during the convectional flow of it. It is also demonstrated here by the Continuous Wavelet Transform (CWT) as well as the Fractal analysis with matlab 7.12 version
A New Enhanced Method of Non Parametric power spectrum Estimation.CSCJournals
The spectral analysis of non uniform sampled data sequences using Fourier Periodogram method is the classical approach. In view of data fitting and computational standpoints why the Least squares periodogram(LSP) method is preferable than the “classical” Fourier periodogram and as well as to the frequently-used form of LSP due to Lomb and Scargle is explained. Then a new method of spectral analysis of nonuniform data sequences can be interpreted as an iteratively weighted LSP that makes use of a data-dependent weighting matrix built from the most recent spectral estimate. It is iterative and it makes use of an adaptive (i.e., data-dependent) weighting, we refer to it as the iterative adaptive approach (IAA).LSP and IAA are nonparametric methods that can be used for the spectral analysis of general data sequences with both continuous and discrete spectra. However, they are most suitable for data sequences with discrete spectra (i.e., sinusoidal data), which is the case we emphasize in this paper. Of the existing methods for nonuniform sinusoidal data, Welch, MUSIC and ESPRIT methods appear to be the closest in spirit to the IAA proposed here. Indeed, all these methods make use of the estimated covariance matrix that is computed in the first iteration of IAA from LSP. MUSIC and ESPRIT, on the other hand, are parametric methods that require a guess of the number of sinusoidal components present in the data, otherwise they cannot be used; furthermore.
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
Design Of Area Delay Efficient Fixed-Point Lms Adaptive Filter For EEG Applic...IJTET Journal
An efficient architecture for the implementation of a delayed least mean square adaptive filter. A Novel
partial product Generator is achieving lower adaptation-delay and Area delay consumption and propose a strategy
for optimized balanced pipelining across the time-consuming combinational blocks of the structure. From synthesis
results, the proposed design will offers less area-delay product (ADP) the best of the existing systolic structures, on
average, for filter lengths N =8, 16, and 32. An efficient fixed-point implementation scheme of the proposed
architecture, The EEG(electroencephalogram) is used for recording of electrical activity of the brain .During
recording the EEG is contaminated by various artifacts as PLI(Power line interference), MA(Muscle artifact),
EBA(Eye blink artifact). This paper gives Detail of various artifacts which occur in EEG signal. In this we study
adaptive filter for reducing the EBA (eye blink artifact) noise from the EEG signal and to increase SNR (Signal to
noise ratio).the analytical result matches with the simulation result is showed.
MODIFIED LLL ALGORITHM WITH SHIFTED START COLUMN FOR COMPLEXITY REDUCTIONijwmn
Multiple-input multiple-output (MIMO) systems are playing an important role in the recent wireless
communication. The complexity of the different systems models challenge different researches to get a good
complexity to performance balance. Lattices Reduction Techniques and Lenstra-Lenstra-Lovàsz (LLL)
algorithm bring more resources to investigate and can contribute to the complexity reduction purposes.
In this paper, we are looking to modify the LLL algorithm to reduce the computation operations by
exploiting the structure of the upper triangular matrix without “big” performance degradation. Basically,
the first columns of the upper triangular matrix contain many zeroes, so the algorithm will perform several
operations with very limited income. We are presenting a performance and complexity study and our
proposal show that we can gain in term of complexity while the performance results remains almost the
same.
Design of an Adaptive Hearing Aid Algorithm using Booth-Wallace Tree MultiplierWaqas Tariq
The paper presents FPGA implementation of a spectral sharpening process suitable for speech enhancement and noise reduction algorithms for digital hearing aids. Booth and Booth Wallace multiplier is used for implementing digital signal processing algorithms in hearing aids. VHDL simulation results confirm that Booth Wallace multiplier is hardware efficient and performs faster than Booth’s multiplier. Booth Wallace multiplier consumes 40% less power compared to Booth multiplier. A novel digital hearing aid using spectral sharpening filter employing booth Wallace multiplier is proposed. The results reveal that the hardware requirement for implementing hearing aid using Booth Wallace multiplier is less when compared with that of a booth multiplier. Furthermore it is also demonstrated that digital hearing aid using Booth Wallace multiplier consumes less power and performs better in terms of speed.
Area efficient parallel LFSR for cyclic redundancy check IJECEIAES
Cyclic Redundancy Check (CRC), code for error detection finds many applications in the field of digital communication, data storage, control system and data compression. CRC encoding operation is carried out by using a Linear Feedback Shift Register (LFSR). Serial implementation of CRC requires more clock cycles which is equal to data message length plus generator polynomial degree but in parallel implementation of CRC one clock cycle is required if a whole data message is applied at a time. In previous work related to parallel LFSR, hardware complexity of the architecture reduced using a technique named state space transformation. This paper presents detailed explaination of search algorithm implementation and technique to find number of XOR gates required for different CRC algorithms. This paper presents a searching algorithm and new technique to find the number of XOR gates required for different CRC algorithms. The comparison between proposed and previous architectures shows that the number of XOR gates are reduced for CRC algorithms which improve the hardware efficiency. Searching algorithm and all the matrix computations have been performed using MATLAB simulations.
Engineering Research Publication
Best International Journals, High Impact Journals,
International Journal of Engineering & Technical Research
ISSN : 2321-0869 (O) 2454-4698 (P)
www.erpublication.org
On selection of periodic kernels parameters in time series predictioncsandit
In the paper the analysis of the periodic kernels parameters is described. Periodic kernels can
be used for the prediction task, performed as the typical regression problem. On the basis of the
Periodic Kernel Estimator (PerKE) the prediction of real time series is performed. As periodic
kernels require the setting of their parameters it is necessary to analyse their influence on the
prediction quality. This paper describes an easy methodology of finding values of parameters of
periodic kernels. It is based on grid search. Two different error measures are taken into
consideration as the prediction qualities but lead to comparable results. The methodology was
tested on benchmark and real datasets and proved to give satisfactory results.
K-Sort: A New Sorting Algorithm that Beats Heap Sort for n 70 Lakhs!idescitation
Sundararajan and Chakraborty [10] introduced a new
version of Quick sort removing the interchanges. Khreisat
[6] found this algorithm to be competing well with some other
versions of Quick sort. However, it uses an auxiliary array
thereby increasing the space complexity. Here, we provide a
second version of our new sort where we have removed the
auxiliary array. This second improved version of the algorithm,
which we call K-sort, is found to sort elements faster than
Heap sort for an appreciably large array size (n 70,00,000)
for uniform U[0, 1] inputs.
IJRET : International Journal of Research in Engineering and Technology is an international peer reviewed, online journal published by eSAT Publishing House for the enhancement of research in various disciplines of Engineering and Technology. The aim and scope of the journal is to provide an academic medium and an important reference for the advancement and dissemination of research results that support high-level learning, teaching and research in the fields of Engineering and Technology. We bring together Scientists, Academician, Field Engineers, Scholars and Students of related fields of Engineering and Technology.
FORTRAN is used as a numerical and scientific computing language. The main objective of the lab work is to understand FORTRAN language using which we solve simple numerical problems and compare different methodologies. Through this project we make use of various functions of FORTRAN and solve a FDM simple heat equation problem applying different conditions viz. Dirichlet and Von Neumann. The given problems are solved analytically then built and compiled using a free integrated development environment called CODE::BLOCKS [1] which is an open source platform for FORTRAN and C.
Using several mathematical examples from three different authors in texts from different courses this paper illustrates the easier way to avoid confusions and always get the correct results with the least effort was to use the proposed Excel Gamma function explained in detail for the proper use of the Q(z) and ercf(x) functions in most communication courses. The paper serves as a tutorial and introduction for such functions
A NEW PARALLEL ALGORITHM FOR COMPUTING MINIMUM SPANNING TREEijscmc
Computing the minimum spanning tree of the graph is one of the fundamental computational problems. In
this paper, we present a new parallel algorithm for computing the minimum spanning tree of an undirected
weighted graph with n vertices and m edges. This algorithm uses the cluster techniques to reduce the
number of processors by fraction 1/f (n) and the parallel work by the fraction O ( 1 lo g ( f ( n )) ),where f (n) is an
arbitrary function. In the case f (n) =1, the algorithm runs in logarithmic-time and use super linear work on
EREWPRAM model. In general, the proposed algorithm is the simplest one.
HEATED WIND PARTICLE’S BEHAVIOURAL STUDY BY THE CONTINUOUS WAVELET TRANSFORM ...cscpconf
Nowadays Continuous Wavelet Transform (CWT) as well as Fractal analysis is generally used for the Signal and Image processing application purpose. Our current work extends the field of application in case of CWT as well as Fractal analysis by applying it in case of the agitated wind particle’s behavioral study. In this current work in case of the agitated wind particle, we have mathematically showed that the wind particle’s movement exhibits the “Uncorrelated” characteristics during the convectional flow of it. It is also demonstrated here by the Continuous Wavelet Transform (CWT) as well as the Fractal analysis with matlab 7.12 version
A New Enhanced Method of Non Parametric power spectrum Estimation.CSCJournals
The spectral analysis of non uniform sampled data sequences using Fourier Periodogram method is the classical approach. In view of data fitting and computational standpoints why the Least squares periodogram(LSP) method is preferable than the “classical” Fourier periodogram and as well as to the frequently-used form of LSP due to Lomb and Scargle is explained. Then a new method of spectral analysis of nonuniform data sequences can be interpreted as an iteratively weighted LSP that makes use of a data-dependent weighting matrix built from the most recent spectral estimate. It is iterative and it makes use of an adaptive (i.e., data-dependent) weighting, we refer to it as the iterative adaptive approach (IAA).LSP and IAA are nonparametric methods that can be used for the spectral analysis of general data sequences with both continuous and discrete spectra. However, they are most suitable for data sequences with discrete spectra (i.e., sinusoidal data), which is the case we emphasize in this paper. Of the existing methods for nonuniform sinusoidal data, Welch, MUSIC and ESPRIT methods appear to be the closest in spirit to the IAA proposed here. Indeed, all these methods make use of the estimated covariance matrix that is computed in the first iteration of IAA from LSP. MUSIC and ESPRIT, on the other hand, are parametric methods that require a guess of the number of sinusoidal components present in the data, otherwise they cannot be used; furthermore.
International Journal of Engineering and Science Invention (IJESI) is an international journal intended for professionals and researchers in all fields of computer science and electronics. IJESI publishes research articles and reviews within the whole field Engineering Science and Technology, new teaching methods, assessment, validation and the impact of new technologies and it will continue to provide information on the latest trends and developments in this ever-expanding subject. The publications of papers are selected through double peer reviewed to ensure originality, relevance, and readability. The articles published in our journal can be accessed online.
Design Of Area Delay Efficient Fixed-Point Lms Adaptive Filter For EEG Applic...IJTET Journal
An efficient architecture for the implementation of a delayed least mean square adaptive filter. A Novel
partial product Generator is achieving lower adaptation-delay and Area delay consumption and propose a strategy
for optimized balanced pipelining across the time-consuming combinational blocks of the structure. From synthesis
results, the proposed design will offers less area-delay product (ADP) the best of the existing systolic structures, on
average, for filter lengths N =8, 16, and 32. An efficient fixed-point implementation scheme of the proposed
architecture, The EEG(electroencephalogram) is used for recording of electrical activity of the brain .During
recording the EEG is contaminated by various artifacts as PLI(Power line interference), MA(Muscle artifact),
EBA(Eye blink artifact). This paper gives Detail of various artifacts which occur in EEG signal. In this we study
adaptive filter for reducing the EBA (eye blink artifact) noise from the EEG signal and to increase SNR (Signal to
noise ratio).the analytical result matches with the simulation result is showed.
MODIFIED LLL ALGORITHM WITH SHIFTED START COLUMN FOR COMPLEXITY REDUCTIONijwmn
Multiple-input multiple-output (MIMO) systems are playing an important role in the recent wireless
communication. The complexity of the different systems models challenge different researches to get a good
complexity to performance balance. Lattices Reduction Techniques and Lenstra-Lenstra-Lovàsz (LLL)
algorithm bring more resources to investigate and can contribute to the complexity reduction purposes.
In this paper, we are looking to modify the LLL algorithm to reduce the computation operations by
exploiting the structure of the upper triangular matrix without “big” performance degradation. Basically,
the first columns of the upper triangular matrix contain many zeroes, so the algorithm will perform several
operations with very limited income. We are presenting a performance and complexity study and our
proposal show that we can gain in term of complexity while the performance results remains almost the
same.
Design of an Adaptive Hearing Aid Algorithm using Booth-Wallace Tree MultiplierWaqas Tariq
The paper presents FPGA implementation of a spectral sharpening process suitable for speech enhancement and noise reduction algorithms for digital hearing aids. Booth and Booth Wallace multiplier is used for implementing digital signal processing algorithms in hearing aids. VHDL simulation results confirm that Booth Wallace multiplier is hardware efficient and performs faster than Booth’s multiplier. Booth Wallace multiplier consumes 40% less power compared to Booth multiplier. A novel digital hearing aid using spectral sharpening filter employing booth Wallace multiplier is proposed. The results reveal that the hardware requirement for implementing hearing aid using Booth Wallace multiplier is less when compared with that of a booth multiplier. Furthermore it is also demonstrated that digital hearing aid using Booth Wallace multiplier consumes less power and performs better in terms of speed.
In this paper, a novel architecture of RNS based 1D Lifting Integer Wavelet Transform (IWT) has been introduced. Advantage of Residue Number System (RNS) based Lifting Scheme over RNS based Filter Bank and non-binary IWT has been discussed. The performance of traditional predicts and updates stage of binary Lifting Scheme (LS) for Discrete Wavelet Transform (DWT) generates huge carry propagation
delay, power and complexity. As a result non binary number system is becoming popular in the field of Digital Signal Processing (DSP) due to its efficient performance. In this paper also a new fixed number ROM based RNS division circuit has been proposed. The proposed architecture has been validated on Xilinx Vertex5 FPGA platform and the corresponding result and reports are shown in here.
Design & Implementation of LUT Based Multiplier Using APCOMS Techniqueijsrd.com
The multiplication is major arithmetic operation in signal processing and in ALU's .The multiplier uses look-up-table (LUT) as memory for their computations. However, we do not find any significant work on LUT optimization for memory-based multiplication. A new approach to LUT design was presented, where only the odd multiple storage (OMS) scheme. In addition to that the antisymmetric product coding (APC) approach, the LUT size is reduced to half and provides a reduction. When APC approach is combined with the OMS technique, the two's complement operations could be simplified since the input address and LUT output could always be transformed into odd integers, and thus reduces the LUT size to one fourth of the conventional LUT. The proposed LUT multipliers for word size L=W=5 bits are coded in VHDL and synthesized in Xilinx 14.2. It is found that the proposed LUT-based multiplier involves comparable area and time complexity for a word size of 5-bits.
Discrete wavelet transform-based RI adaptive algorithm for system identificationIJECEIAES
In this paper, we propose a new adaptive filtering algorithm for system identifica- tion. The algorithm is based on the recursive inverse (RI) adaptive algorithm which suffers from low convergence rates in some applications; i.e., the eigenvalue spread of the autocorrelation matrix is relatively high. The proposed algorithm applies discrete-wavelet transform (DWT) to the input signal which, in turn, helps to overcome the low convergence rate of the RI algorithm with relatively small step-size(s). Different scenarios has been investigated in different noise environments in system identification setting. Experiments demonstrate the advantages of the proposed DWT recursive inverse (DWT-RI) filter in terms of convergence rate and mean-square-error (MSE) compared to the RI, discrete cosine transform LMS (DCT-LMS), discretewavelet transform LMS (DWT-LMS) and recursive-least-squares (RLS) algorithms under same conditions.
DISTRIBUTION LOAD FLOW ANALYSIS FOR RDIAL & MESH DISTRIBUTION SYSTEMIAEME Publication
Power flow analysis is the backbone of power system analysis and design. They are necessary for planning, operation, economic scheduling and exchange of power between utilities. Power flow analysis is required for many other analyses such as transient stability, optimal power flow and contingency studies. The principal information of power flow analysis is to find the magnitude and phase angle of voltage at each bus and the real and reactive power flowing in each transmission lines. Power flow analysis is an importance tool involving numerical analysis applied to a power system. In this analysis, iterative techniques are used due to there no known analytical method to solve the problem. This resulted nonlinear set of equations or called power flow equations are generated.
Adaptive Channel Equalization using Multilayer Perceptron Neural Networks wit...IOSRJVSP
This research addresses the problem inter-symbol interference (ISI) using equalization techniques for time dispersive channels with additive white Gaussian noise (AWGN). The channel equalizer is modelled as a non-linear Multilayer Perceptron (MLP) structure. The Back Propagation (BP) algorithm is used to optimize the synaptic weights of the equalizer during the training mode. In the typical BP algorithm, the error signal is propagated from the output layer to the input layer while the learning rate parameter is held constant. In this study, the BP algorithm is modified so as to allow for the learning rate to be variable at each iteration and this achieves a faster convergence. The proposed algorithm is used to train the MLP based decision feedback equalizer (DFE) for time dispersive ISI channels. The equalizer is tested for a random input sequence of BPSK signals and its performance analysed in terms of the Bit Error Rates and speed of convergence. Simulation results show that the proposed algorithm improves the Bit Error Rate (BER) and rate of convergence.
Multi-carrier Equalization by Restoration of RedundancY (MERRY) for Adaptive ...IJNSA Journal
This paper proposes a new blind adaptive channel shortening approach for multi-carrier systems. The performance of the discrete Fourier transform-DMT (DFT-DMT) system is investigated with the proposed DST-DMT system over the standard carrier serving area (CSA) loop1. Enhanced bit rates demonstrated and less complexity also involved by the simulation of the DST-DMT system.
Performance Assessment of Polyphase Sequences Using Cyclic Algorithmrahulmonikasharma
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Low Power Adaptive FIR Filter Based on Distributed Arithmetic
1. S Ramanathan et al. Int. Journal of Engineering Research and Applications www.ijera.com
ISSN: 2248-9622, Vol. 6, Issue 5, (Part - 7) May 2016, pp.47-51
www.ijera.com 47 | P a g e
Low Power Adaptive FIR Filter Based on Distributed Arithmetic
S Ramanathan1
, Gorty Anand1
, Prasanth Reddy1
, Prof. Sri Adibhatla Sridevi2
1
( M.Tech, VLSI Design, School of Electronics Engineering (SENSE), VIT University, Vellore)
2
(Associate Professor, Micro & Nanoelectronics, School of Electronics Engineering (SENSE), VIT University,
Vellore)
ABSTRACT
This paper aims at implementation of a low power adaptive FIR filter based on distributed arithmetic (DA) with
low power, high throughput, and low area. Least Mean Square (LMS) Algorithm is used to update the weight
and decrease the mean square error between the current filter output and the desired response. The pipelined
Distributed Arithmetic table reduces switching activity and hence it reduces power. The power consumption is
reduced by keeping bit-clock used in carry-save accumulation much faster than clock of rest of the operations.
We have implemented it in Quartus II and found that there is a reduction in the total power and the core dynamic
power by 31.31% and 100.24% respectively when compared with the architecture without DA table.
Keywords: Adaptive filter, Distributed Arithmetic, Finite Impulse Response, Least Mean Square algorithm,
Lookup Table
I. INTRODUCTION
An adaptive filter tries to model the
relationship between two real time signals using an
iterative approach [1].
Four aspects defines an adaptive filter:
1) The input signals of the filter
2) The impulse response of the filter
3) The filter coefficients
4) The adaptive algorithm that is used to adjust the
weights
The LMS algorithm given by Widrow-Hoff
is used to update the tapped-delay line FIR filter's
weights because of its simplicity and convergence
performance provided by it [2].
DA based technique without multipliers [3],
are used as they provide high-throughput processing
capability and regularity that results in computing
structures that are cost-effective and area-time
efficient [4].
II. THE ADAPTIVE FILTER
In the block diagram shown in Fig.1 a
sample of x(n), the input signal, is fed to adaptive
filter and the filter outputs y(n). This output is then
compared with d(n), the desired response, and the
difference of the two is the error signal e(n) as
shown in (1).
e (n) = d (n) - y (n) (1)
Fig. 1. The general adaptive filter
Fig. 2. System identification using adaptive filter
The error signal is fed to the adaptive filter
which, in a well-defined manner, updates the
coefficients of the filter from time n to time (n + 1).
Through this adaptation process, as n increases, the
magnitude of e (n) decreases, or in other words, the
output of the adaptive filter tends to the desired
response signal.
Let x (n) be the input to an unknown
system and let dˆ (n) be its corresponding output.
Then, the desired response signal is given by
d (n) = dˆ (n) + η (n) (2)
The role of adaptive filter here is to
precisely represent dˆ (n) at its output. It can be
said that the adaptive filter driven by x (n), has
accurately modeled the unknown system if y (n) = dˆ
(n).
III. LMS ADAPTIVE ALGORITHM
In every cycle, the LMS algorithm
calculates an output and the corresponding error
value. It then uses the estimated error to update the
weights in every cycle. The LMS adaptive filter
weights in nth iteration are updated as per the
following equations.
w(n + 1) = w(n) + μ.e(n).x(n) (3a)
RESEARCH ARTICLE OPEN ACCESS
2. S Ramanathan et al. Int. Journal of Engineering Research and Applications www.ijera.com
ISSN: 2248-9622, Vol. 6, Issue 5, (Part - 7) May 2016, pp.47-51
www.ijera.com 48 | P a g e
Where
e(n) = d(n) - y(n) (3b)
y(n) = wT
(n).x(n) (3c)
At the nth iteration, x(n) the input vector
and w(n) the weight vector, are respectively given
by,
x(n) = [x(n), x(n - 1),…,x(n - N + 1)]T
(4a)
w(n) = [w0(n),w1(n),…,wN-1(n)]T
(4b)
In the above equations, d(n) denotes the
desired response, y(n) denotes the filter's output in
the nth iteration, e(n) is the error computed in the nth
iteration and it is used for updating the weights, μ is
the convergence-factor and N is the length of the
filter.
In pipelined designs, only after certain
number of cycles, the feedback error e(n) becomes
available, this deley is known as the ―adaptation-
delay‖. Hence the pipelined architectures use e(n -
m) which is the delayed error in place of recent-most
error to update the current weight. Here m denotes
the adaptation-delay. The following equation is the
weight-update equation of such delayed LMS
adaptive filter
w(n + 1) = w(n) +μ.e(n - m).x(n - m) (5)
IV. DA-BASED APPROACH FOR
INNER-PRODUCT COMPUTATION
In every cycle, the LMS adaptive filter has
to compute an inner-product and this contributes to
the most of the critical-path. Let the inner-product of
(3c) be given by (6) for simplicity of presentation,
and is in fact the arithmetic sum of products which
defines the response of linear time-invariant (LTI)
network.
N-1
y = ∑ wk.xk (6)
k = 0
Where xk and wk form the N-point vectors
corresponding the recent-most N-1 input and current
weights respectively for 0 ≤ k ≤ N-1. Assuming the
bit-width of the weight to be L, each component of
weight vector can be expressed in 2’s complement
representation as shown in (7).
L-1
wk = - wk0 + ∑ wkl.2-l
(7)
l = 1
where wkl is the lth bit of wk. Substituting (7) in
(6), we get
N-1 L-1
y = ∑ xk [- wk0 + ∑ wkl.2-l
] (8)
k = 0 l = 1
Fig. 3. Conventional DA-based implementation of 4-
point inner-product.
rewriting (8), we get
N N L-1
y = - ∑ wk0.xk + ∑ ∑ wkl.xk.2-l
(9)
k = 1 k = 1 l = 1
expanding (9), we get (10)
y = - [w10.x1 + w20.x2 + w30.x3 + … + wk0.xk]
+ [w11.x1 + w21.x2 + w31.x3 + … + wk1.xk] 2-1
+ [w12.x1 + w22.x2 + w32.x3 + … + wk2.xk] 2-2
:
+ [w1(B-2).x1 + w2(B-2).x2 + … + wk(B-2).xk] 2-(B-2)
+ [w1(B-1).x1 + … +wk(B-1).xk] 2-(B-1)
(10)
Every term inside the brackets is actually a
binary AND operation that involves all the bits of
the constant and one bit of the input and the plus
signs are arithmetic sum operations. An LUT can
now be constructed which can be addressed by the
same scaled bit of all the input variables and can
access the sum of the terms inside each pair of
brackets. Fig.3 shows the LUT.
Converting the sum-of-products form of
(10) into a distributed form, we get (11)
Fig. 4. Carry-save implementation of the shift-
accumulation.
3. S Ramanathan et al. Int. Journal of Engineering Research and Applications www.ijera.com
ISSN: 2248-9622, Vol. 6, Issue 5, (Part - 7) May 2016, pp.47-51
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Fig. 5. DA-table for generation of the possible
sums of the input samples.
N-1 N-1 L-1
y = - ∑ wk0.xk + ∑ 2-l
. [ ∑ xk.wkl ] (11)
k = 0 k = 0 l = 1
the inner-product given by (11) is computed as
L-1 N-1
y = [ ∑ 2-l
.yl ] - y0 where yl = ∑ xk.wkl (12)
l = 1 k = 1
The partial-sum yl for l = 0, 1,2,…,L-1, can
have 2N
possible values, as the elements of the N-
point bit-sequence {wkl for 0 ≤ k ≤ N - 1} can be
either 0 or 1.
Fig. 6. The structure of DA-based LMS adaptive
filter of length N = 4.
Now, using the bit-sequence {wkl} as
address bits for the computation of inner-product, we
can read out the partial sums yl from the LUT, if we
precompute all the 2N
possible values of yl and store
it in the LUT.
Therefore, inner-product of (12) can be
calculated be calculated in L cycles of shift-
accumulation which is followed by the LUT read
operations corresponding to L number of bit-slices
{wkl} for 0 ≤ l ≤ L-1. This is shown in Fig.3. The
shift-accumulation is performed using carry-save
accumulator, as the shift-accumulation shown in
Fig.3 involves significant critical-path. This is
shown in Fig.4. The bit-slices of vector w are given
to the carry-save accumulator one after the other in
the order LSB to MSB. But in case of MSB slices,
the negative or the 2’s complement of the output of
the LUT must to be accumulated. This could easily
be achieved using XOR gates. Therefore, all the bits
of LUT output are fed to XOR gates with sign-
control input. If MSB slice appears as address, then
alone the sign-control is set to 1. So the XOR gates
produce the 1’s complement of the LUT output if
MSB slice appears as address and does not affect the
output for other cases. Lastly, the sum and carry
words that are obtained after L clock cycles are
added by an adder and the input carry of this adder is
set to 1 to account for the 2’s complement operation
of the LUT output for the MSB slice.
Fig. 7. (a) Structure of 4-point inner-product block.
(b) Structure of weight increment block for N = 4.
(c) The logic which is used for the generation of the
control word t for the barrel-shifter for L = 8.
Fig. 8. Structure of DA-based LMS AF with N =
16 and P = 4.
The content of kth LUT location is expressed as
N-1
ck = ∑ xj.kj (13)
j = 0
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ISSN: 2248-9622, Vol. 6, Issue 5, (Part - 7) May 2016, pp.47-51
www.ijera.com 50 | P a g e
where kj is (j + 1)th bit of the N-bit binary
representation of integer k where k lies in the range
0 ≤ k ≤ 2N
- 1. We can pre-compute ck for 0 ≤ k ≤ 2N
- 1 and store it in a RAM-based LUT of 2N
words.
But here we store (2N
- 1) words in a DA-table that
consists of 2N
- 1 registers, in place of storing 2N
words in LUT. Fig.5 shows an example of one such
DA-table for N = 4. It has only 15 registers that are
used to store the sums of input words which are pre-
computed. Seven adders in parallel, compute seven
new values of ck.
V. DA BASED ADAPTIVE FILTER
STRUCTURE
Fig.6 shows the structure of DA-based adaptive filter
of filter length N = 4. It has a 4-point inner-product
and a block for weight-increment, along with the
additional circuits required for the computation of
e(n), the error value and the control word t for
barrel-shifters.
As shown in Fig.7a, the 4-point inner-product block
consists of a DA-table that has an array of 15
registers that store the partial-inner-products yl for l
in range 0 < l ≤ 15 and to select the content of one of
these registers, a 16 : 1 MUX which is used.
Bit-slices of weights A = {w3l w2l w1l w0l}
for 0 ≤ l ≤ L - 1 are given to the MUX as control in
the order LSB to MSB, and the output of MUX is
given to carry-save accumulator which is shown in
Fig. 4.
The carry-save accumulator shift-
accumulates all the partial-inner-products after L bit-
cycles, and generates two words of size (L + 2)-bit
each during these L bit-cycles, one for sum and other
one for the carry. The sum and carry words are
shifted-added with an input carry ’1’ in order to
generate filter output which is subtracted from d(n),
the desired output, later to obtain the error value
e(n).
By assuming N = PQ, we can now
decompose inner-product computation of (6) into N
= P small adaptive filtering blocks of length P as
shown in (14)
P-1 2P-1 N-1
y = ∑ wk.xk + ∑ wk.xk + … + ∑ wk.xk
(14)
k = 0 k = P k = N-P
Each of the above mentioned P-point inner-
product computation blocks, to update P weights,
have weight-increment unit. The structure for N = 16
and P = 4 is shown in the Fig.8. It has four inner-
product blocks of length P = 4 as shown in Fig.7a.
Two separate binary adder-trees are used to add the
(L + 2)-bit carry and sum produced by these four
blocks. Four carry-in bits are added to the sum
words which are the output of four 4-point inner-
product blocks. At the first level binary adder-tree of
the carry words, two carry-in bits are set as the input
carry, as carry words are of twice the weight as
compared to sum words. This is same as inclusion of
four carry-in to sum words.
To make the length of sign-magnitude
separator as L-bit, assuming μ = 1/N, we can
truncate the 4 LSBs of error e(n) for N = 16. The
performance of adaptive filter is not very much
affected by the truncation as the design
requires the location of most significant 1 of μ.e(n).
VI. RESULTS
The Low Power Adaptive FIR Filter Based
on Distributed Arithmetic has been implemented in
5. S Ramanathan et al. Int. Journal of Engineering Research and Applications www.ijera.com
ISSN: 2248-9622, Vol. 6, Issue 5, (Part - 7) May 2016, pp.47-51
www.ijera.com 51 | P a g e
Quartus II and we have compared it with an
architecture without distributed arithmetic table.
TABLE I. POWER CONSUMPTION
Architecture Power Consumption
Total Power Core Dynamic Power
Conventional
FIR Filter
110.77mW 24.91mW
DA-Based
LMS Adaptive
FIR Filter
84.36mW 12.44mW
It is found that there is a reduction in the
total power and the core dynamic power by 31.31%
and 100.24% respectively when compared with the
architecture without DA table.
VII. CONCLUSION
We have implemented an efficient
pipelined architecture for high-throughput, low-
power and low-area DA-based adaptive filter.
Throughput rate is quite significantly improved by
means of concurrent processing of weight-update
and filtering operation and the parallel LUT update.
For computation of the filter output, we have used a
carry-save accumulation unit for signed partial-
inner-products computation.
ACKNOWLEDGEMENTS
We have implemented a Low Power
Adaptive FIR Filter Based on Distributed Arithmetic
under the supervision of Prof. Sri Adibhatla Sridevi,
Associate Professor, SENSE. We thank her for
providing proper guidance and timely help which
helped us to successfully complete the work.
REFERENCES
Books:
[1]. Douglas S.C. Introduction to Adaptive
Filters, Digital Signal Processing
Handbook, Ed. Vijay K. Madisetti and
Douglas B. Williams, Boca Raton: CRC
Press LLC, 1999.
[2]. S. Haykin and B. Widrow, Least-mean-
square adaptive filters. Wiley-Interscience,
Hoboken, NJ, 2003.
Journal Papers:
[3]. S. A. White, ―Applications of the
distributed arithmetic to digital signal
processing: A tutorial review,‖ IEEE ASSP
Magazine, vol. 6, no. 3, pp. 5–19, Jul. 1989.
[4]. S. Y. Park And P. K. Meher, "Low-Power,
High-Throughput, and Low-Area Adaptive
FIR Filter Based On DA" IEEE
Transactions on Circuits And Systems—II,
Express Briefs, vol. 60, no. 6, June 2013,
pp. 346-350.