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.
Improved Timing Estimation Using Iterative Normalization Technique for OFDM S...IJECEIAES
Conventional timing estimation schemes based on autocorrelation experience perfor- mance degradation in the multipath channel environment with high delay spread. To overcome this problem, we proposed an improvement of the timing estimation for the OFDM system based on statistical change of symmetrical correlator. The new method uses iterative normalization technique to the correlator output before the detection based on statistical change of symmetric correlator is applied. Thus, it increases the detection probability and achieves better performance than previously published methods in the multipath environment. Computer simulation shows that our method is very robust in the fading multipath channel.
Optimization of Collective Communication in MPICH Lino Possamai
This is a lecture about the paper: "Optimization of Collective Communication in MPICH". Department of Computer Science, University Ca' Foscari of Venice, Italy
BER Analysis ofImpulse Noise inOFDM System Using LMS,NLMS&RLSiosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
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.
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.
Improved Timing Estimation Using Iterative Normalization Technique for OFDM S...IJECEIAES
Conventional timing estimation schemes based on autocorrelation experience perfor- mance degradation in the multipath channel environment with high delay spread. To overcome this problem, we proposed an improvement of the timing estimation for the OFDM system based on statistical change of symmetrical correlator. The new method uses iterative normalization technique to the correlator output before the detection based on statistical change of symmetric correlator is applied. Thus, it increases the detection probability and achieves better performance than previously published methods in the multipath environment. Computer simulation shows that our method is very robust in the fading multipath channel.
Optimization of Collective Communication in MPICH Lino Possamai
This is a lecture about the paper: "Optimization of Collective Communication in MPICH". Department of Computer Science, University Ca' Foscari of Venice, Italy
BER Analysis ofImpulse Noise inOFDM System Using LMS,NLMS&RLSiosrjce
IOSR Journal of Computer Engineering (IOSR-JCE) is a double blind peer reviewed International Journal that provides rapid publication (within a month) of articles in all areas of computer engineering and its applications. The journal welcomes publications of high quality papers on theoretical developments and practical applications in computer technology. Original research papers, state-of-the-art reviews, and high quality technical notes are invited for publications.
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.
On Channel Estimation of OFDM-BPSK and -QPSK over Nakagami-m Fading ChannelsCSCJournals
This paper evaluates the performance of OFDM - BPSK & -QPSK based system with and without channel estimation over Nakagami-m fading channels. Nakagami-m variants are generated by decomposition of Nakagami random variable into orthogonal random variables with Gaussian distribution envelopes. Performance of OFDM system in Nakagami channel has been reported here. The results yield the optimum value of m based on BER and SNR. Using this optimum value of m, Channel estimation over flat fading has been reported here. It has been depicted clearly from simulated graphs that channel estimation has further reduced the BER. However, threshold value of m has played a vital role during channel estimation.
A Review: Compensation of Mismatches in Time Interleaved Analog to Digital Co...IJERA Editor
The execution of today's correspondence frameworks is exceedingly subject to the utilized Analog-to-Digital converters (ADCs), and with a specific end goal to give more flexibility and exactness to the developing correspondence innovations, superior-ADCs are needed. In this respect, the time-interleaved operation of an exhibit of ADCs (TI-ADC) might be a sensible result. A TI-ADC can build its throughput by utilizing M channel ADCs or sub converters in parallel and examining the data motion in a period-interleaved way. In any case, the execution of a TI-ADC gravely suffers from the bungles around the channel ADCs. In this paper we survey the advancement in the configuration of low-intricacy advanced remedy structures and calculations for time-interleaved ADCs in the course of the most recent five years. We devise a discrete-time model, state the outline issue, and finally infer the calculations and structures. Specifically, we examine proficient calculations to outline time-differing remedy filters and additionally iterative structures using polynomial based filters. Thusly, the remuneration structure may be utilized to repay time-differing recurrence reaction befuddles in time-interleaved ADCs, and in addition to remake uniform examples from nonuniformly tested indicators. We examine the recompense structure, research its execution, and exhibit requisition zones of the structure through various illustrations. At long last, we give a standpoint to future examination questions.
A Novel CAZAC Sequence Based Timing Synchronization Scheme for OFDM SystemIJAAS Team
Several classical timing synchronization schemes have been proposed for the timing synchronization in OFDM systems based on the correlation between identical parts of OFDM symbol. These schemes show poor performance due to the presence of plateau and significant side lobe. In this paper we present a timing synchronization schemes with timing metric based on a Constant Amplitude Zero Auto Correlation (CAZAC) sequence. The performance of the proposed timing synchronization scheme is better than the classical techniques.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Review on Doubling the Rate of SEFDM Systems using Hilbert Pairsijtsrd
A novel multi carrier technique for spectrally efficient frequency division multiplexing SEFDM system for improving the spectral efficiency is discussed. A Hilbert pair is utilized as pulse shaping filters. At the the Hilbert pulse pair is generated using the square root raised cosine pulse and an equivalent matched filter configuration is utilized to generate the Hilbert pair at receiver. Simulations with different values of compression factor of the SEFDM signals were carried out to verify the data rate gain of the proposed system. The proposed system has no degradation in bit error rate performance with the data rate doubled relative to conventional SEFDM system. For system using turbo coding, there is significant BER improvement compared to uncoded transmission. Padmam Kaimal "Review on Doubling the Rate of SEFDM Systems using Hilbert Pairs" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-5 | Issue-4 , June 2021, URL: https://www.ijtsrd.compapers/ijtsrd42348.pdf Paper URL: https://www.ijtsrd.comengineering/electronics-and-communication-engineering/42348/review-on-doubling-the-rate-of-sefdm-systems-using-hilbert-pairs/padmam-kaimal
Flexible dsp accelerator architecture exploiting carry save arithmeticNexgen Technology
2016 ieee project ,2016-2017 ieee projects, application projects, best ieee projects, bulk final year projects, bulk ieee projects ,diploma projects electrical engineering electrical engineering projects ,final year application projects, final year csc projects, final year cse project, final year it projects ,final year project, final year projects, final year projects in chennai ,final year projects in coimabtore, final year projects in hyderabad, final year projects in pondicherry final year projects in rajasthan ,ieee based projects for ece, ieee final year projects, ieee master, ieee project, ieee project 2015 ,ieee project 2016, ieee project centers in pondicherry ,ieee project for eee, ieee projects, ieee projects ,2015-2016 ieee projects, 2016-2017 ieee projects, cse ieee projects, cse 2015 ieee projects, cse 2016 ieee projects for cse ,ieee projects for it, ieee projects in bangalore, ieee projects in chennai, ieee projects in coimbatore, ieee projects in hyderabad ,ieee projects in madurai ,ieee projects in maharashtra ,ieee projects in mumbai, ieee projects in odisha, ieee projects in orissa, ieee projects in pondicherry, ieee projects in pondy ,ieee projects in pune, ieee projects in uttarakhand, ieee projects titles, 2015-2016 latest projects for eee, NEXGEN TECHNOLOGY mtech ieee projects mtech projects 2016-2017 mtech projects in chennai mtech, projects in cuddalore ,mtech projects in neyveli, mtech projects in panruti, mtech projects in pondicherry, mtech projects in tindivanam, mtech projects in villupuram, online ieee projects ,phd guidance, project for engineering ,project titles for ece
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.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
Reverberation is a well known effect particularly important for listening of recorded and live music. In this paper we propose a real implementation of an enhanced approach for digital artificial reverberator. Starting from a preliminary analysis of the mixing time, the selected impulse response is decomposed in the time domain considering the early and late reflections. Therefore, a short FIR is used to synthesize the first part of the impulse response, and a generalized recursive structure is used to synthesize the late reflections, exploiting a minimization criterion in the cepstral domain. Several results are reported taking into consideration different real impulse responses and comparing the results with those obtained with previous techniques in terms of computational complexity and reverberation quality.
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.
On Channel Estimation of OFDM-BPSK and -QPSK over Nakagami-m Fading ChannelsCSCJournals
This paper evaluates the performance of OFDM - BPSK & -QPSK based system with and without channel estimation over Nakagami-m fading channels. Nakagami-m variants are generated by decomposition of Nakagami random variable into orthogonal random variables with Gaussian distribution envelopes. Performance of OFDM system in Nakagami channel has been reported here. The results yield the optimum value of m based on BER and SNR. Using this optimum value of m, Channel estimation over flat fading has been reported here. It has been depicted clearly from simulated graphs that channel estimation has further reduced the BER. However, threshold value of m has played a vital role during channel estimation.
A Review: Compensation of Mismatches in Time Interleaved Analog to Digital Co...IJERA Editor
The execution of today's correspondence frameworks is exceedingly subject to the utilized Analog-to-Digital converters (ADCs), and with a specific end goal to give more flexibility and exactness to the developing correspondence innovations, superior-ADCs are needed. In this respect, the time-interleaved operation of an exhibit of ADCs (TI-ADC) might be a sensible result. A TI-ADC can build its throughput by utilizing M channel ADCs or sub converters in parallel and examining the data motion in a period-interleaved way. In any case, the execution of a TI-ADC gravely suffers from the bungles around the channel ADCs. In this paper we survey the advancement in the configuration of low-intricacy advanced remedy structures and calculations for time-interleaved ADCs in the course of the most recent five years. We devise a discrete-time model, state the outline issue, and finally infer the calculations and structures. Specifically, we examine proficient calculations to outline time-differing remedy filters and additionally iterative structures using polynomial based filters. Thusly, the remuneration structure may be utilized to repay time-differing recurrence reaction befuddles in time-interleaved ADCs, and in addition to remake uniform examples from nonuniformly tested indicators. We examine the recompense structure, research its execution, and exhibit requisition zones of the structure through various illustrations. At long last, we give a standpoint to future examination questions.
A Novel CAZAC Sequence Based Timing Synchronization Scheme for OFDM SystemIJAAS Team
Several classical timing synchronization schemes have been proposed for the timing synchronization in OFDM systems based on the correlation between identical parts of OFDM symbol. These schemes show poor performance due to the presence of plateau and significant side lobe. In this paper we present a timing synchronization schemes with timing metric based on a Constant Amplitude Zero Auto Correlation (CAZAC) sequence. The performance of the proposed timing synchronization scheme is better than the classical techniques.
International Journal of Engineering Research and Applications (IJERA) is an open access online peer reviewed international journal that publishes research and review articles in the fields of Computer Science, Neural Networks, Electrical Engineering, Software Engineering, Information Technology, Mechanical Engineering, Chemical Engineering, Plastic Engineering, Food Technology, Textile Engineering, Nano Technology & science, Power Electronics, Electronics & Communication Engineering, Computational mathematics, Image processing, Civil Engineering, Structural Engineering, Environmental Engineering, VLSI Testing & Low Power VLSI Design etc.
Review on Doubling the Rate of SEFDM Systems using Hilbert Pairsijtsrd
A novel multi carrier technique for spectrally efficient frequency division multiplexing SEFDM system for improving the spectral efficiency is discussed. A Hilbert pair is utilized as pulse shaping filters. At the the Hilbert pulse pair is generated using the square root raised cosine pulse and an equivalent matched filter configuration is utilized to generate the Hilbert pair at receiver. Simulations with different values of compression factor of the SEFDM signals were carried out to verify the data rate gain of the proposed system. The proposed system has no degradation in bit error rate performance with the data rate doubled relative to conventional SEFDM system. For system using turbo coding, there is significant BER improvement compared to uncoded transmission. Padmam Kaimal "Review on Doubling the Rate of SEFDM Systems using Hilbert Pairs" Published in International Journal of Trend in Scientific Research and Development (ijtsrd), ISSN: 2456-6470, Volume-5 | Issue-4 , June 2021, URL: https://www.ijtsrd.compapers/ijtsrd42348.pdf Paper URL: https://www.ijtsrd.comengineering/electronics-and-communication-engineering/42348/review-on-doubling-the-rate-of-sefdm-systems-using-hilbert-pairs/padmam-kaimal
Flexible dsp accelerator architecture exploiting carry save arithmeticNexgen Technology
2016 ieee project ,2016-2017 ieee projects, application projects, best ieee projects, bulk final year projects, bulk ieee projects ,diploma projects electrical engineering electrical engineering projects ,final year application projects, final year csc projects, final year cse project, final year it projects ,final year project, final year projects, final year projects in chennai ,final year projects in coimabtore, final year projects in hyderabad, final year projects in pondicherry final year projects in rajasthan ,ieee based projects for ece, ieee final year projects, ieee master, ieee project, ieee project 2015 ,ieee project 2016, ieee project centers in pondicherry ,ieee project for eee, ieee projects, ieee projects ,2015-2016 ieee projects, 2016-2017 ieee projects, cse ieee projects, cse 2015 ieee projects, cse 2016 ieee projects for cse ,ieee projects for it, ieee projects in bangalore, ieee projects in chennai, ieee projects in coimbatore, ieee projects in hyderabad ,ieee projects in madurai ,ieee projects in maharashtra ,ieee projects in mumbai, ieee projects in odisha, ieee projects in orissa, ieee projects in pondicherry, ieee projects in pondy ,ieee projects in pune, ieee projects in uttarakhand, ieee projects titles, 2015-2016 latest projects for eee, NEXGEN TECHNOLOGY mtech ieee projects mtech projects 2016-2017 mtech projects in chennai mtech, projects in cuddalore ,mtech projects in neyveli, mtech projects in panruti, mtech projects in pondicherry, mtech projects in tindivanam, mtech projects in villupuram, online ieee projects ,phd guidance, project for engineering ,project titles for ece
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.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
Reverberation is a well known effect particularly important for listening of recorded and live music. In this paper we propose a real implementation of an enhanced approach for digital artificial reverberator. Starting from a preliminary analysis of the mixing time, the selected impulse response is decomposed in the time domain considering the early and late reflections. Therefore, a short FIR is used to synthesize the first part of the impulse response, and a generalized recursive structure is used to synthesize the late reflections, exploiting a minimization criterion in the cepstral domain. Several results are reported taking into consideration different real impulse responses and comparing the results with those obtained with previous techniques in terms of computational complexity and reverberation quality.
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.
FOLDED ARCHITECTURE FOR NON CANONICAL LEAST MEAN SQUARE ADAPTIVE DIGITAL FILT...VLSICS Design
Power consumption reduction is transpiring drift in area of VLSI digital signal processing. This gives rise
to need of minimization of silicon area which is done by folding algorithm. As silicon area decreases power
consumption of a circuit decreases. Folding is an algorithm which reduces silicon chip area by combining
various arithmetic operations into one operation by time scheduling technique. It is applied on iterative
data flow graph with appropriate folding set. Least mean square algorithm alters coefficients of Adaptive
filter in order to achieve desired output. Proposed work is focused on design of efficient VLSI architecture
for LMS adaptive filter aims at reducing mainly area which results in power consumption reduction and
hardware complexity. LMS filter structure used here is called non-canonical as transpose FIR structure is
used. Results show that numbers of adders are reduced by 37.5 % and multipliers by 33.33% without
changing characteristics of filter.
Low Power Adaptive FIR Filter Based on Distributed ArithmeticIJERA Editor
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
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.
International Journal of Engineering Research and Development (IJERD)IJERD Editor
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yahoo journals, bing journals, International Journal of Engineering Research and Development, google journals, hard copy of journal
Performance Analysis of OFDM Transceiver with Folded FFT and LMS Filteridescitation
This paper presents an Orthogonal Frequency Division Multiplexing (OFDM)
transceiver that makes use of a low power Fast Fourier Transform (FFT) along with a Least
Mean Square (LMS) filter. The folded FFT is developed via folding transformation and
register minimization techniques with real values as inputs which leads to reduction in
hardware complexity by exploiting the redundancy present in computing the FFT samples
and also the amount of power consumed. A LMS filter is also designed for the purpose of
noise removal. The OFDM transceiver with the folded FFT and LMS filter is analyzed in
terms of error performance to validate the advantages of less power consumption and
hardware utilization when compared to the traditional OFDM system with conventional
FFT. The individual components and the entire OFDM system that has been proposed are
modeled using Verilog HDL and functionally verified using Xilinx ISIM simulator.
Optimized implementation of an innovative digital audio equalizera3labdsp
Digital audio equalization is one of the most common operations in the acoustic field, but its performance
depends on computational complexity and filter design techniques. Starting from a previous FIR
implementation based on multirate systems and filterbanks theory, an optimized digital audio equalizer
is derived. The proposed approach employs IIR filters to improve the filterbanks structure developed to
avoid ripple between adjacent bands. The effectiveness of the optimized implementation is shown comparing
it with the previous approach. The solution presented here has several advantages increasing
the equalization performance in terms of low computational complexity, low delay, and uniform frequency
response.
Time domain analysis and synthesis using Pth norm filter designCSCJournals
In this paper, a new approach for the design and implementation of FIR filter banks for multirate analysis and synthesis is explored. The method is based on the least algorithm and takes into consideration the characteristics of the individual filters. Features of the proposed approach include; it does not need to adapt the weighting function involved and no constraints are imposed during the course of optimization. Mostly, the FIR filter design is concentrated around linear phase characteristics but with the help of minimax solution for FIR filters using the least- algorithm, this optimal filter design approach helps us to enhance the properties of LTI systems with better stability filter coefficient convergence. Hence norm algorithm will be used in multirate to explore the stability and other properties. We have proposed the band analysis system for analysis and synthesis purpose to explore multirate filter banks. The Matlab toolbox has been used for implementing the filters and its properties will be verified with various plots and tables. The results of this paper enable us to achieve good signal to noise ratio with analysis and synthesis level operations.
Design of Optimized FIR Filter Using FCSD Representation IJEEE
This paper presents the design and implementation of an eight order efficient FIR filter for wireless communication system. In this work, factored canonical signed digit representation (FCSD) is used for representing the filter coefficients in order to reduce the design complexity, area and delay of the FIR filter. Complexity of the system has been reduced by replacing binary coefficients with FCSD representation. Further area and delay has been improved by replacing multiplication operation with add and shift method where carry save adder (CSA) is used for addition of two numbers and barrel shifter is used for shifting the data words. Representation of coefficient in the FCSD format along with fastest adder and shifter improves the performance of the system. FIR filter has been designed using an equiripple method in MATLAB and further synthesized on Spartan 3E XC3S500E target FPGA device. Simulation results show that optimized FCSD based FIR filter offers a less number of slices, look up tables (LUTs) and flip-flops as compared to CSD and conventional FCSD based FIR filter, in addition to enhanced performance.
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.
Comparative Analysis of Different Wavelet Functions using Modified Adaptive F...IJERA Editor
The traditional method of wavelet denoising is inefficient in removing the overlap noise between noisy signal
and noise, due to which a modified adaptive filtering based on wavelet transform method is introduced. The
method used in this paper filters out the noise on the basis of wavelet denoising using different wavelet
functions. The simulation results indicate the Signal to Noise ratio (SNR), Mean Square Error (MSE) and signal
error power spectral density comparison plot between different wavelet functions. These comparison results
verified that Daubechies is more efficient than other wavelet functions in filtering out noise in all perspectives.
Modified Adaptive Lifting Structure Of CDF 9/7 Wavelet With Spiht For Lossy I...idescitation
We present a modified structure of 2-D cdf 9/7 wavelet
transforms based on adaptive lifting in image coding. Instead
of alternately applying horizontal and vertical lifting, as in
present practice, Adaptive lifting performs lifting-based
prediction in local windows in the direction of high pixel
correlation. Hence, it adapts far better to the image orientation
features in local windows. The predicting and updating signals
of Adaptive lifting can be derived even at the fractional pixel
precision level to achieve high resolution, while still
maintaining perfect reconstruction. To enhance the
performance of adaptive based modified structure of 2-D CDF
9/7 is coupled with SPIHT coding algorithm to improve the
drawbacks of wavelet transform. Experimental results show
that the proposed modified scheme based image coding
technique outperforms JPEG 2000 in both PSNR and visual
quality, with the improvement up to 6.0 dB than existing
structure on images with rich orientation features .
Design and Implementation of a Programmable Truncated Multiplierijsrd.com
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D0341015020
1. International Journal of Engineering Science Invention
ISSN (Online): 2319 – 6734, ISSN (Print): 2319 – 6726
www.ijesi.org Volume 3 Issue 4 ǁ April 2014 ǁ PP.15-20
www.ijesi.org 15 | Page
An Efficient Adaptive Fir Filter Based On Distributed Arithmetic
1,
M.Usha , 2,
R.Ramadoss
1,
P.G.Scholar , 2,
Assistant Professor
1,2
,Dept. of Electronics and communication Engineering Sri Muthukumaran Institute Of Technology
Chennai,Tamilnadu,India
ABSTRACT: Adaptive filtering constitutes an important class of DSP algorithms employed in several hand
held mobile devices for applications such as echo cancellation, signal de-noising, and channel equalization.
The throughput of the proposed design is increased by parallel lookup table (LUT) update.The 16:1 multiplexer
is replace by a 8:1 and 2:1 MUX. The conventional adder-based shift accumulation for DA-based inner-product
computation is replaced by conditional signed carry-save accumulation in order to reduce the area complexity;
the power consumption of the proposed design is reduced by using a fast bit clock for all operations. It involves
the same number of multiplexors, smaller LUT, and nearly half the number of adders compared to the existing
DA-based design. The proposed architecture is found to involve significantly 29% less area-13% less power
and throughput compared with the existing DA-based implementations of FIR filter and a increase in operating
frequency of 12MHZ is achieved.
KEYWORDS: Adaptive filter, circuit optimization, distributed arithmetic (DA), least mean square (LMS)
algorithm
I. INTRODUCTION
Adaptive filters are widely used in several digital signal processing (DSP) applications. The tap-delay
line finite impulse response (FIR) filters whose weights are updated by the famous Widrow-Hoff least mean
square (LMS) algorithm [1] is the most popularly used adaptive filter not only due to its simplicity but also due
to its satisfactory convergence performance [2]. The direct form FIR filter configuration for the implementation
of LMS adaptive filter results in either zero or lower adaptation-delay but involves a large critical-path due to an
inner-product computation to obtain the filter output. Therefore, when the input signal has high sample-rate, the
critical-path could exceed the sample period. In such cases, it is necessary to reduce the critical-path by
pipelined implementation. Since the conventional LMS algorithm does not support pipelined implementation
due to its recursive behavior, it is modified to a form called delayed LMS algorithm [3], which allows pipelined
implementation of different sections of the adaptive filter. State-of-the-art hardware implementation of adaptive
filters typically involves DSP microprocessors or custom logic design using one or more hardware multiply-
accumulate (MAC) units. While an implementation employing DSP microprocessors provides easy
programmability, the serial implementation on a single processing unit adversely affects the throughput of these
filters. This is especially true for higher order filters. Custom logic design using one or more hardware MAC
units may be used to parallelize the implementation and thus improve the throughput but at the cost of increased
logic complexity, chip area usage, and power consumption [4].
Various types of DSP operations are employed in practice. Filtering is one of the most widely used
signal processing operations [5]. For FIR filters, output y(n) is a linear convolution of weights wn and inputs.
Distributed arithmetic (DA) is one way to implement convolution without multiplier, where the MAC
operations are replaced by a series of LUT access and summations. Techniques, such as ROM decomposition
[6] and offset-binary coding (OBC) [7] can reduce the LUT size, which would otherwise increase exponentially
with the filter lengthN+1for conventional DA. However, in many applications such as echo cancelation and
system identification, coefficient adaptation is needed. This adaptation makes it challenging to implement DA-
based adaptive filters with low cost due to the necessity of updating LUTs. Several approaches have been
developed for DA-based adaptive filters, i.e., from the point of view of reducing logic complexity [8]. Recently,
a DA-based FIR adaptive filter implementation scheme has been presented in [9], which uses extra “auxiliary”
LUTs to help in the updating; however, memory usage is doubled. The rest of this paper is organized as follows.
Section 2 describes the Review of LMS Adaptive Algorithms. Section 3 introduces the proposed architecture
and Proposed DA based approach for inner product computation, Section 4 describes the simulation results for
proposed system. Conclusions are finally drawn in Section 5.
2. An Efficient Adaptive Fir Filter Based…
www.ijesi.org 16 | Page
II. EXISTING SYSTEM
A. Review of LMS Adaptive Algorithms
During each cycle, the LMS algorithm computes a filter output and an error value that is equal to the
difference between the current filter output and the desired response. The estimated error is then used to update
the filter weights in every training cycle. The weights of LMS adaptive filter during the nth iteration are updated
according to the following equations:
w(n+1) = w(n) + μ·e(n)·x(n) (1)
where
e(n) = d(n) − y(n) (2)
y(n) = w qT(n)·x(n). (3)
The input vector x(n)and the weight vector w(n)at the nth training iteration are respectively given by
x(n = [x(n),x(n−1),...,x(n−N+1)]T (4)
w(n)= [w0(n),w1(n),...,wN−1(n)]T. (5)
d(n) is the desired response, and y(n) is the filter output of the nth iteration. e(n)denotes the error computed
during the nth iteration, which is used to update the weights, μ is the convergence factor, and N is the filter
length.In the case of pipelined designs, the feedback error e(n) becomes available after certain number of
cycles, called the “adaptation delay.” The pipelined architectures therefore use the delayed error e(n−m) for
updating the current weight instead of the most recent error, where m is the adaptation de-lay. The weight-
update equation of such delayed LMS adaptive filter is given by
w(n+1)= w(n)+μ·e(n−m)·x(n−m). (6)
Fig. 1. Existing System Block Diagram
III. PROPOSED SYSTEM
A. Proposed DA based Adaptive Filter
The proposed structure of DA-based adaptive filter of length N=4 is shown in Fig. 2. It consists of a
four-point inner-product block and a weight-increment block along with additional circuits for the computation
of error value e(n) and control word t for the barrel shifters. The four-point inner-product block Fig. 3 includes
a DA table consisting of an array of 15 registers which stores the partial inner products yl for 0 < l ≤ 15and a 16
: 1 multiplexor (MUX) to select the content of one of those registers. Bit slices of weights A={w3l w2l w1l
w0l} for 0 ≤ l ≤ L−1are fed to the MUX as control in LSB-to-MSB order, and the output of the MUX is fed to
the carry-save accumulator (shown in Fig. 2). After L bit cycles, the carry-save accumulator shift accumulates
all the partial inner products and generates a sum word and a carry word of size (L+2) bit each. The carry and
sum words are shifted added with an input carry “1” to generate filter output which is subsequently subtracted
from the desired output d(n)to obtain the error e(n). The magnitude of the computed error is decoded to generate
the control word t for the barrel shifter. The logic used for the generation of control word t to be used for the
barrel shifter. The convergence factor μ is usually taken to be O(1/N). We have taken μ= 1/N. However, one can
take μ as 2 −i/N, where i is a small integer. The number of shifts t in that case is increased by i, and the input to
the barrel shifters is pre shifted by i locations accordingly to reduce the hardware complexity.
3. An Efficient Adaptive Fir Filter Based…
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Fig. 2. Proposed System Structure
The weight-increment unit consists of four barrel shifters and four adder/subtractor cells. The barrel
shifter shifts the different input values xk for k= 0,1,...,N−1 by appropriate number of locations (deter-mined by
the location of the most significant one in the estimated error). The barrel shifter yields the desired increments
to be added with or subtracted from the current weights. The sign bit of the error is used as the control for
adder/subtractor cells such that, when sign bit is zero or one, the barrel-shifter output is respectively added with
or subtracted from the content of the corresponding current value in the weight register.
B. Proposed DA based approach for inner product computation
The LMS adaptive filter, in each cycle, needs to perform an inner-product computation which
contributes to the most of the critical path. For simplicity of presentation, let the inner product of (3) be given
by
Where Wk and XK for 0≤ k ≤ N−1 form the N-point vectors corresponding the current weights and most recent
N−1 input, respectively. Assuming L to be the bit width of the weight, each component of the weight vector
may be expressed in two’s complement representation
Where Wkl denotes the lth
bit of Wk. Substituting (8), we can write (7) in an expanded form
To convert the sum-of-products form of (7) into a distributed form, the order of summations over the indices k
and l in (9) can be interchanged to have
(10)
and the inner product given by (10) can be computed as
where
4. An Efficient Adaptive Fir Filter Based…
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Since any element of the N-point bit sequence {wkl for 0≤ k≤N−1}can either be zero or one, the partial sum yl
for l=0,1,...,L−1can have possible values. If all the possible values of yl are precomputed and stored in a
LUT, the partial sums yl can be read out from the LUT using the bit sequence{wkl}as address bits for
computing the inner product.The inner product can be calculated in L cycles of shift accumulation, followed by
LUT-read operations corresponding to L number of bit slices{wkl}for 0≤ l ≤ L−1.The bit slices of vector ware
fed one after the next in the least significant bit (LSB) to the most significant bit (MSB) order to the carry-save
accumulator. However, the negative (two’s complement) of the LUT output needs to be accumulated in case of
MSB slices. Therefore, all the bits of LUT output are passed through XOR gates with a sign-control input which
is set to one only when the MSB slice appears complement of the LUT output corresponding to the MSB slice
but do not affect the output for other bit slices. Finally, the sum and carry words obtained after L clock cycles
are required to be added by a final adder (not shown in the figure), and the input carry of the final adder is
required to be set to one to account for the two’s complement operation of the LUT output corresponding to the
MSB slice.
The content of the kth
LUT location can be expressed as
(12)
where kj is the (j+1)th
bit of kth
LUT location can be expressed as N-bit binary representation of integer k for
0≤k≤ −1. Note that ck for 0≤k≤ −1 can be pre computed and stored in RAM-based LUT of words.
Fig. 3. . DA table for generation of possible sums of input samples.
However, instead of storing words in LUT, an example of such a DA table for N=4 is shown in Fig. 4. It
contains only 15 registers to store the as address. The XOR gates thus produce the one’s pre computed sums of
input words. Seven new values of ck are computed by seven adders in parallel.
Fig. 4 Internal blocks of inner product computation block.
5. An Efficient Adaptive Fir Filter Based…
www.ijesi.org 19 | Page
IV. RESULTS AND DISCUSSION
The proposed System was executed on Windows XP operating system at an operating frequency of
2.0GHz using Xillinx simu lator.
Fig. 5. Proposed method pin diagram without slow clock
Fig. 6. Filter output wave form
Its seen from the synthesis result that the frequency of the proposed system is increased by 12 MHZ.
V. CONCLUSION
We have suggested an efficient pipelined architecture for low-power, high-throughput, and low-area
implementation of DA-based adaptive filter. Throughput rate is significantly enhanced by parallel LUT update
and concurrent processing of filtering operation and weight-update operation. We have also proposed a carry-
save accumulation scheme of signed partial inner products for the computation of filter output. From the
synthesis results, we find that the proposed design consumes 13% less power and 29% less ADP over our
previous DA-based FIR adaptive filter in average for filter lengths N=16 Compared to the best of other existing
designs, our proposed architecture provides 9.5 times less power and 4.6 times less ADP. Offset binary coding
is popularly used to reduce the LUT size to half for area-efficient implementation of DA which can be applied
to our design as well.
6. An Efficient Adaptive Fir Filter Based…
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REFERENCES
[1] B. Widrow and S. D. Stearns, Adaptive signal processing. Prentice Hall, Englewood Cliffs, NJ, 1985.
[2] S. Haykin and B. Widrow, Least-mean-square adaptive filters. Wiley-Interscience, Hoboken, NJ, 2003.
[3] M. Meyer and P. Agrawal, “A modular pipelined implementation of a delayed LMS transversal adaptive filter,” in Proc. IEEE
International Symposium on Circuits and Systems, ISCAS ’09, May 1990, pp. 1943–1946.
[4] D. Allred, V. Krishnan, W. Huang, and D. Anderson, “Implementation of an LMS adaptive filter on an FPGA employing
multiplexed multiplier architecture,” in Proc. Asilomar Conf. Signals Systems Computers, Nov. 2003, pp.918-921.
[5] S. K. Mitra, Digital Signal Processing: A Computer-Based Approach, 2nd
. New York: McGraw-Hill, 2001.
[6] K.K.Parhi, VLSI Digital Signal Processing Systems: Design and Implementation. Hoboken, NJ: Wiley, 1999.
[7] S. A. White, “Applications of distributed arithmetic to digital signal processing: A tutorial review,”IEEE ASSP Mag., vol. 6, no.
3, pp. 4–19,Jul. 1989.
[8] D. J. Allred, H. Yoo, V. Krishnan, W. Huang, and D. V. Anderson, “An FPGA implementation for a high throughput adaptive
filter using distributed arithmetic,” in Proc. 12th Annu. IEEE Symp. Field-Programmable Custom Comput. Mach., 2004, pp.
324–325
[9] D. J. Allred, H. Yoo, V. Krishnan, W. Huang, and D. V. Anderson, “A novel high performance distributed arithmetic adaptive
filter implementation on an FPGA,” in Proc. IEEE Int. Conf. Acoust., Speech, Signal Process., 2004, vol. 5, pp. V-161–V-164