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.
Capsulization of Existing Space Time TechniquesIJEEE
1) The document discusses space-time coding techniques used in wireless communication systems to improve reliability of data transmission using multiple transmit antennas.
2) It describes space-time block codes (STBC) such as Alamouti codes and orthogonal designs which transmit redundant copies of data across antennas without loss of data rate.
3) It also discusses space-time trellis codes (STTC) which provide coding gain but have higher complexity than STBCs.
Efficient FPGA implementation of high speed digital delay for wideband beamfor...journalBEEI
In this paper, the authors present an FPGA implementation of a digital delay for beamforming applications. The digital delay is based on a Parallel Farrow Filter. Such architecture allows to reach a very high processing rate with wideband signals and it is suitable to be used with Time-Interleaved Analog to Digital Converters (TI-ADC). The proposed delay has been simulated in MATLAB, implemented on FPGA and characterized in terms of amplitude and phase response, maximum clock frequency and area.
IMPROVEMENT OF LTE DOWNLINK SYSTEM PERFORMANCES USING THE LAGRANGE POLYNOMIAL...IJCNCJournal
The document describes research on improving the performance of LTE downlink systems using Lagrange polynomial interpolation for channel estimation. It presents the MIMO-OFDM transmission scheme used in LTE and discusses various channel estimation techniques including linear, sinus cardinal, Newton polynomial, and Lagrange polynomial interpolation. Simulation results show that Lagrange polynomial interpolation outperforms other methods in terms of block error rate, throughput, and error vector magnitude versus signal-to-noise ratio. The optimal order of the Lagrange polynomial is determined by evaluating performance for different orders.
A Novel Approach of Area-Efficient FIR Filter Design Using Distributed Arithm...IOSR Journals
Abstract: In this paper, a highly area-efficient multiplier-less FIR filter is presented. Distributed Arithmetic (DA) has been used to implement a bit-serial scheme of a general asymmetric version of an FIR filter, taking optimal advantage of the 3-input LUT-based structure of FPGAs. The implementation of FIR filters on FPGA based on traditional arithmetic method costs considerable hardware resources, which goes against the decrease of circuit scale and the increase of system speed. This paper presents the realization of area efficient architectures using Distributed Arithmetic (DA) for implementation of Finite Impulse Response (FIR) filter. The performance of the bit-serial and bit parallel DA along with pipelining architecture with different quantized versions are analyzed for FIR filter Design. Distributed Arithmetic structure is used to increase the resource usage while pipeline structure is also used to increase the system speed. In addition, the divided LUT method is also used to decrease the required memory units. However, according to Distributed Arithmetic, we can make a Look-Up-Table (LUT) to conserve the MAC values and callout the values according to the input data if necessary. Therefore, LUT can be created to take the place of MAC units so as to save the hardware resources. The simulation results indicate that FIR filters using Distributed Arithmetic can work stable with high speed and can save almost 50 percent hardware resources to decrease the circuit scale, and can be applied to a variety of areas for its great flexibility and high reliability. This method not only reduces the LUT size, but also modifies the structure of the filter to achieve high speed performance. Keywords: DSP, Digital Filters, FIR , FPGA, MAC, Distributed Arithmetic(DA),Divided LUT, pipeline
Comparison of the link budget with experimental performance of a wi max systemPfedya
The document compares the link budget calculations of a WiMAX system to experimental performance measurements in a suburban environment. It describes the configuration of the WiMAX system operating at 3.5 GHz, including the base station parameters, receiver sensitivity, and scenario. It then presents the methodology, which involves measuring path loss to develop a path loss model, using that model to calculate link budgets and expected ranges for different modulations, and validating these by taking actual performance measurements with WiMAX modems. The results of the path loss measurements and link budget calculations are then compared to the experimental performance measurements to validate the link budget approach.
This document describes a simulation of an optical orthogonal frequency division multiplexing (OFDM) system using different M-ary quadrature amplitude modulation (QAM) techniques. The performance of the system was evaluated in terms of bit error rate (BER) and signal-to-noise ratio (SNR) by varying the modulation order M to 16, 64, and 256. Simulation results showed that higher modulation orders require higher SNR values to achieve the same BER as lower modulation orders. The simulated results closely matched theoretical predictions, though there was a deviation of 3-3.5 dB, likely due to additional noise and losses not accounted for in the theoretical model.
Ofdm sim-matlab-code-tutorial web for EE studentsMike Martin
This document describes an OFDM simulation using Matlab. It begins with an introduction to OFDM and its advantages for wireless communications. It then provides the mathematical equations for OFDM transmission and reception based on the DVB-T standard. The document outlines the steps to simulate OFDM transmission in Matlab, including generating OFDM symbols using an IFFT, adding a guard interval, pulse shaping, and upconverting to a carrier frequency. It also provides the equations and steps for simulating OFDM reception. Figures and tables are included to illustrate the simulation results and parameters.
Capsulization of Existing Space Time TechniquesIJEEE
1) The document discusses space-time coding techniques used in wireless communication systems to improve reliability of data transmission using multiple transmit antennas.
2) It describes space-time block codes (STBC) such as Alamouti codes and orthogonal designs which transmit redundant copies of data across antennas without loss of data rate.
3) It also discusses space-time trellis codes (STTC) which provide coding gain but have higher complexity than STBCs.
Efficient FPGA implementation of high speed digital delay for wideband beamfor...journalBEEI
In this paper, the authors present an FPGA implementation of a digital delay for beamforming applications. The digital delay is based on a Parallel Farrow Filter. Such architecture allows to reach a very high processing rate with wideband signals and it is suitable to be used with Time-Interleaved Analog to Digital Converters (TI-ADC). The proposed delay has been simulated in MATLAB, implemented on FPGA and characterized in terms of amplitude and phase response, maximum clock frequency and area.
IMPROVEMENT OF LTE DOWNLINK SYSTEM PERFORMANCES USING THE LAGRANGE POLYNOMIAL...IJCNCJournal
The document describes research on improving the performance of LTE downlink systems using Lagrange polynomial interpolation for channel estimation. It presents the MIMO-OFDM transmission scheme used in LTE and discusses various channel estimation techniques including linear, sinus cardinal, Newton polynomial, and Lagrange polynomial interpolation. Simulation results show that Lagrange polynomial interpolation outperforms other methods in terms of block error rate, throughput, and error vector magnitude versus signal-to-noise ratio. The optimal order of the Lagrange polynomial is determined by evaluating performance for different orders.
A Novel Approach of Area-Efficient FIR Filter Design Using Distributed Arithm...IOSR Journals
Abstract: In this paper, a highly area-efficient multiplier-less FIR filter is presented. Distributed Arithmetic (DA) has been used to implement a bit-serial scheme of a general asymmetric version of an FIR filter, taking optimal advantage of the 3-input LUT-based structure of FPGAs. The implementation of FIR filters on FPGA based on traditional arithmetic method costs considerable hardware resources, which goes against the decrease of circuit scale and the increase of system speed. This paper presents the realization of area efficient architectures using Distributed Arithmetic (DA) for implementation of Finite Impulse Response (FIR) filter. The performance of the bit-serial and bit parallel DA along with pipelining architecture with different quantized versions are analyzed for FIR filter Design. Distributed Arithmetic structure is used to increase the resource usage while pipeline structure is also used to increase the system speed. In addition, the divided LUT method is also used to decrease the required memory units. However, according to Distributed Arithmetic, we can make a Look-Up-Table (LUT) to conserve the MAC values and callout the values according to the input data if necessary. Therefore, LUT can be created to take the place of MAC units so as to save the hardware resources. The simulation results indicate that FIR filters using Distributed Arithmetic can work stable with high speed and can save almost 50 percent hardware resources to decrease the circuit scale, and can be applied to a variety of areas for its great flexibility and high reliability. This method not only reduces the LUT size, but also modifies the structure of the filter to achieve high speed performance. Keywords: DSP, Digital Filters, FIR , FPGA, MAC, Distributed Arithmetic(DA),Divided LUT, pipeline
Comparison of the link budget with experimental performance of a wi max systemPfedya
The document compares the link budget calculations of a WiMAX system to experimental performance measurements in a suburban environment. It describes the configuration of the WiMAX system operating at 3.5 GHz, including the base station parameters, receiver sensitivity, and scenario. It then presents the methodology, which involves measuring path loss to develop a path loss model, using that model to calculate link budgets and expected ranges for different modulations, and validating these by taking actual performance measurements with WiMAX modems. The results of the path loss measurements and link budget calculations are then compared to the experimental performance measurements to validate the link budget approach.
This document describes a simulation of an optical orthogonal frequency division multiplexing (OFDM) system using different M-ary quadrature amplitude modulation (QAM) techniques. The performance of the system was evaluated in terms of bit error rate (BER) and signal-to-noise ratio (SNR) by varying the modulation order M to 16, 64, and 256. Simulation results showed that higher modulation orders require higher SNR values to achieve the same BER as lower modulation orders. The simulated results closely matched theoretical predictions, though there was a deviation of 3-3.5 dB, likely due to additional noise and losses not accounted for in the theoretical model.
Ofdm sim-matlab-code-tutorial web for EE studentsMike Martin
This document describes an OFDM simulation using Matlab. It begins with an introduction to OFDM and its advantages for wireless communications. It then provides the mathematical equations for OFDM transmission and reception based on the DVB-T standard. The document outlines the steps to simulate OFDM transmission in Matlab, including generating OFDM symbols using an IFFT, adding a guard interval, pulse shaping, and upconverting to a carrier frequency. It also provides the equations and steps for simulating OFDM reception. Figures and tables are included to illustrate the simulation results and parameters.
IRJET - Co-Axial Fed Tri-Slot Antenna for Triple-Band ApplicationIRJET Journal
This document describes the simulation and design of a coaxial fed tri-band microstrip patch antenna for wireless communication applications. The antenna is designed to operate in three bands: 2.28-2.35 GHz, 4.05-4.11 GHz, and 5.8-5.9 GHz. Rectangular slots are introduced on the patch to achieve the triple band performance. The dimensions of the slots and ground plane as well as the feed position are optimized using HFSS simulator. Simulation results show return losses less than -25 dB across the three bands and VSWR less than 2, indicating good impedance matching between the feed line and patch. Radiation patterns and efficiencies over 95% are obtained at the center frequencies
Performance Evaluation of H.264 AVC Using CABAC Entropy Coding For Compound I...DR.P.S.JAGADEESH KUMAR
The document evaluates the performance of H.264/AVC entropy coding (CABAC) for compressing different types of images. It is observed that CABAC is highly efficient at compressing compound images, achieving higher compression ratios and PSNR values compared to other image types at high bitrates. The proposed system compresses grayscale compound images using CABAC after applying Daubechies wavelet transform. Performance is measured using compression ratio and PSNR metrics at varying bits per pixel.
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
1) The document proposes a robust audio watermarking technique based on spread spectrum and empirical mode decomposition (EMD). EMD decomposes audio signals into intrinsic mode functions (IMFs) representing different frequencies.
2) The technique embeds a watermark by performing discrete cosine transform (DCT) on extrema points of the last IMF, arranging them in descending order, and adding watermark bits. Extraction retrieves the watermark from lower IMFs using the secret key.
3) Experiments show the technique is robust against attacks like noise, cropping, filtering and wiener filtering.
1) The document discusses techniques for reducing the peak-to-average power ratio (PAPR) of orthogonal frequency division multiplexing (OFDM) signals, including repeated clipping and filtering (RCF) and nonlinear commanding transform (NCT).
2) RCF projects clipping noise into the feasible extension area while removing out-of-band interference through filtering, but suffers from problems like in-band distortion, peak re-growth, and out-of-band radiation.
3) NCT aims to reduce PAPR by compressing peak signals and expanding small signals while maintaining constant average power through optimal parameter selection. The proposed NCT technique outperforms RCF in terms of lower clipping ratio,
Image transmission in ofdm using m ary psk modulation schemes –a comparitive ...eSAT Journals
This document summarizes a study that compares the performance of different modulation schemes for transmitting images over an OFDM system under noisy channel conditions. It is found that BPSK and QPSK modulation schemes perform better than 16-PSK and 256-PSK for image transmission when the signal-to-noise ratio is less than 12dB, with lower bit error rates and fewer errors in received pixel values. For higher SNR scenarios, 16-PSK provides higher data rates but requires SNR above 12dB for acceptable image quality. 256-PSK has the highest data rate but very high bit error rates below SNR of 60dB, resulting in poor received image quality.
Analysis of Phase Noise and Gaussian Noise in terms of Average BER for DP 16-...IRJET Journal
This document analyzes phase noise and Gaussian noise in terms of average bit error rate (BER) for a 112 Gbps dual polarization 16-QAM (DP 16-QAM) optical coherent receiver using digital signal processing (DSP) and different digital filters. It first describes the DP 16-QAM coherent receiver system and the DSP techniques used, including carrier phase estimation. It then simulates the system using Optisystem and MATLAB software and analyzes the phase noise before carrier phase estimation and Gaussian noise after by plotting average BER versus optical signal-to-noise ratio for various filter types and orders. The results show that the 3rd order Gaussian filter provided the lowest average BER and therefore the best noise performance.
The document summarizes research on energy efficient hybrid precoding for simultaneous wireless information and power transfer (SWIPT)-enabled massive MIMO non-orthogonal multiple access (NOMA) systems. It proposes a novel approach using an analog circuit-embedded (ACE) algorithm to design the analog precoder based on a switch-incorporated (SI) architecture. The ACE algorithm uses a probabilistic model and smoothed updating procedure to generate the analog precoding matrix with elements of ±1/√N. Simulation results show the proposed method achieves higher spectral and energy efficiency compared to existing techniques.
A Peak to Average Power Ratio (PAPR) Reduction in OFDM SystemsIRJET Journal
This document discusses peak-to-average power ratio (PAPR) reduction techniques for orthogonal frequency division multiplexing (OFDM) systems. It begins with an introduction to OFDM and the problem of high PAPR values in OFDM signals. It then describes the clipping and filtering method and parabolic peak cancellation method for PAPR reduction. It analyzes these techniques by evaluating complementary cumulative distribution function (CCDF) curves and bit error rate (BER) with the goal of minimizing PAPR while maintaining acceptable BER. Power amplifier nonlinearity is also discussed as a key factor affected by high PAPR OFDM signals.
This document discusses the effects of time offset (TO) and carrier frequency offset (CFO) on orthogonal frequency division multiplexing (OFDM) systems and proposes interference cancellation techniques. It introduces the OFDM system model and how CFO causes loss of orthogonality between subcarriers, resulting in inter-carrier interference (ICI) that degrades performance. The paper proposes an ICI reduction scheme using self-cancellation and evaluates its performance compared to standard OFDM. It also discusses using space-time block coding (STBC) with OFDM to improve performance by reducing bit error rate (BER) under different signal-to-noise ratios. Simulation results show STBC effectively mitigates the effects of inter-
This paper proposes a novel design for a high-speed six-transistor full adder using a two-transistor XOR gate to reduce power dissipation and area. Previous full adder designs used more transistors, resulting in higher power consumption and area. The proposed design uses a two-transistor XOR gate as a building block for an eight-transistor full adder. Simulation results show the new design has lower power consumption and transistor count compared to previous designs.
Compression: Video Compression (MPEG and others)danishrafiq
This document provides an overview of video compression techniques used in standards like MPEG and H.261. It discusses how uncompressed video data requires huge storage and bandwidth that compression aims to address. It explains that lossy compression methods are needed to achieve sufficient compression ratios. The key techniques discussed are intra-frame coding using DCT and quantization similar to JPEG, and inter-frame coding using motion estimation and compensation to remove temporal redundancy between frames. Motion vectors are found using techniques like block matching and sum of absolute differences. MPEG and other standards use a combination of these intra and inter-frame coding techniques to efficiently compress video for storage and transmission.
Simulation of ofdm modulation adapted to the transmission of a fixed imageIAEME Publication
This document summarizes a study on using OFDM modulation adapted for transmitting a fixed image over a disturbed channel. OFDM modulation was simulated using MATLAB. Both classic OFDM with a 25% guard interval and modified OFDM with a reduced guard interval below 25% were evaluated. Results were presented for various M-PSK modulation formats including BPSK, QPSK, 16PSK and 256PSK. Convolutional coding was also used to improve transmission quality. The performance of the system was evaluated in terms of the visual quality of the received image and parameters like SNR, BER under different modulation schemes and guard interval durations. Transmission of up to 98% of the original image quality was achieved.
Point Sum Average Peak Algorithm Detection of LTEAM Publications
This paper proposes an improved detection algorithm for LTE Random Access preamble detection and evaluates the performance of the algorithm with respect to the performance of algorithms proposed in the literature using MDP metric.
Cooperative partial transmit sequence for papr reduction in space frequency b...IAEME Publication
This document discusses a proposed Cooperative Partial Transmit Sequence (Co-PTS) technique for reducing Peak-to-Average Power Ratio (PAPR) in Space Frequency Block Code (SFBC) Multiple-Input Multiple-Output Orthogonal Frequency Division Multiplexing (MIMO-OFDM) signals. The proposed Co-PTS technique combines alternate optimization and spatial sub-block circular permutation. Alternate optimization reduces computational complexity while spatial sub-block circular permutation increases the number of candidate sequences, improving PAPR reduction performance. Simulation results show the proposed Co-PTS technique achieves a lower PAPR of 4.7dB compared to previous PAPR reduction techniques for MIMO-OFDM and SF
Comparative Analysis of DP QPSK and DP 16-QAM Optical Coherent Receiver, with...IRJET Journal
This document compares DP QPSK and DP 16-QAM optical coherent receivers in terms of average bit error rate (BER) when analyzing phase noise. It simulates a 112 Gbps DP 16-QAM and DP QPSK coherent receiver system with digital signal processing (DSP) using Optisystem and MATLAB. The analysis introduces noise before the receiver by varying the optical signal-to-noise ratio (OSNR) and measures average BER. Graphs of average BER versus OSNR are produced for different digital filters and filter orders to determine the filter with minimum phase noise. The DP 16-QAM system shows better power spectrum confinement and is analyzed in more detail.
This paper aims, a 3D-Pilot Aided Multi-Input Multi-Output Orthogonal Frequency Division Multiplexing (MIMO-OFDM) Channel Estimation (CE) for Digital Video Broadcasting -T2 (DVB-T2) for the 5 different proposed block and comb pilot patterns model and performed on different antenna configuration. The effects of multi-transceiver antenna on channel estimation are addressed with different pilot position in frequency, time and the vertical direction of spatial domain framing. This paper first focus on designing of 5-different proposed spatial correlated pilot pattern model with optimization of pilot overhead. Then it demonstrates the performance comparison of Least Square (LS) & Linear Minimum Mean Square Error (LMMSE), two linear channel estimators for 3D-Pilot Aided patterns on different antenna configurations in terms of Bit Error Rate. The simulation results are shown for Rayleigh fading noise channel environments. Also, 3x4 MIMO configuration is recommended as the most suitable configuration in this noise channel environments.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
This paper proposes an enhanced hybrid reverberator that uses both measured impulse responses and synthesized impulse responses to model reverberation effects. It develops an automatic procedure to set the parameters of the hybrid reverberator by analyzing the mixing time of measured impulse responses and minimizing a loss function in the cepstral domain. Experimental results show the proposed method produces higher quality reverberation effects than a previous method, with only a small increase in computational cost, as confirmed by listening tests.
A NOVEL APPROACH TO CHANNEL DECORRELATION FOR STEREO ACOUSTIC ECHO CANCELLATI...a3labdsp
This document proposes a novel approach to decorrelating stereo acoustic signals for acoustic echo cancellation based on the psychoacoustic phenomenon of the "missing fundamental". The approach tracks and removes the pitch from one channel of the stereo signal using an adaptive notch filter, which greatly reduces inter-channel coherence in the lower spectrum without affecting signal quality. Experimental results show the proposed approach provides significant coherence reduction and faster convergence speed of adaptive filters compared to a masked noise injection method, while better preserving the stereo quality.
Low Power High-Performance Computing on the BeagleBoard Platforma3labdsp
The ever increasing energy requirements of supercomputers and server farms is driving the scientific and industrial communities to take in deeper consideration the energy efficiency of computing equipments. This contribution addresses the issue proposing a cluster of ARM processors for high-performance computing. The cluster is composed of five BeagleBoard-xM, with one board managing the cluster, and the other boards executing the actual processing. The software platform is based on the Angstrom GNU/Linux distribution and is equipped with a distributed file system to ease sharing data and code among the nodes of the cluster, and with tools for managing tasks and monitoring the status of each node. The computational capabilities of the cluster have been assessed through High-Performance Linpack and a cluster-wide speaker diarization algorithm, while power consumption has been measured using a clamp meter. Experimental results obtained in the speaker diarization task showed that the energy efficiency of the BeagleBoard-xM cluster is comparable to the one of a laptop computer equipped with a Intel Core2 Duo T8300 running at 2.4 GHz. Furthermore, removing the bottleneck due to the Ethernet interface, the BeagleBoard-xM cluster is able to achieve a superior energy efficiency.
IRJET - Co-Axial Fed Tri-Slot Antenna for Triple-Band ApplicationIRJET Journal
This document describes the simulation and design of a coaxial fed tri-band microstrip patch antenna for wireless communication applications. The antenna is designed to operate in three bands: 2.28-2.35 GHz, 4.05-4.11 GHz, and 5.8-5.9 GHz. Rectangular slots are introduced on the patch to achieve the triple band performance. The dimensions of the slots and ground plane as well as the feed position are optimized using HFSS simulator. Simulation results show return losses less than -25 dB across the three bands and VSWR less than 2, indicating good impedance matching between the feed line and patch. Radiation patterns and efficiencies over 95% are obtained at the center frequencies
Performance Evaluation of H.264 AVC Using CABAC Entropy Coding For Compound I...DR.P.S.JAGADEESH KUMAR
The document evaluates the performance of H.264/AVC entropy coding (CABAC) for compressing different types of images. It is observed that CABAC is highly efficient at compressing compound images, achieving higher compression ratios and PSNR values compared to other image types at high bitrates. The proposed system compresses grayscale compound images using CABAC after applying Daubechies wavelet transform. Performance is measured using compression ratio and PSNR metrics at varying bits per pixel.
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
1) The document proposes a robust audio watermarking technique based on spread spectrum and empirical mode decomposition (EMD). EMD decomposes audio signals into intrinsic mode functions (IMFs) representing different frequencies.
2) The technique embeds a watermark by performing discrete cosine transform (DCT) on extrema points of the last IMF, arranging them in descending order, and adding watermark bits. Extraction retrieves the watermark from lower IMFs using the secret key.
3) Experiments show the technique is robust against attacks like noise, cropping, filtering and wiener filtering.
1) The document discusses techniques for reducing the peak-to-average power ratio (PAPR) of orthogonal frequency division multiplexing (OFDM) signals, including repeated clipping and filtering (RCF) and nonlinear commanding transform (NCT).
2) RCF projects clipping noise into the feasible extension area while removing out-of-band interference through filtering, but suffers from problems like in-band distortion, peak re-growth, and out-of-band radiation.
3) NCT aims to reduce PAPR by compressing peak signals and expanding small signals while maintaining constant average power through optimal parameter selection. The proposed NCT technique outperforms RCF in terms of lower clipping ratio,
Image transmission in ofdm using m ary psk modulation schemes –a comparitive ...eSAT Journals
This document summarizes a study that compares the performance of different modulation schemes for transmitting images over an OFDM system under noisy channel conditions. It is found that BPSK and QPSK modulation schemes perform better than 16-PSK and 256-PSK for image transmission when the signal-to-noise ratio is less than 12dB, with lower bit error rates and fewer errors in received pixel values. For higher SNR scenarios, 16-PSK provides higher data rates but requires SNR above 12dB for acceptable image quality. 256-PSK has the highest data rate but very high bit error rates below SNR of 60dB, resulting in poor received image quality.
Analysis of Phase Noise and Gaussian Noise in terms of Average BER for DP 16-...IRJET Journal
This document analyzes phase noise and Gaussian noise in terms of average bit error rate (BER) for a 112 Gbps dual polarization 16-QAM (DP 16-QAM) optical coherent receiver using digital signal processing (DSP) and different digital filters. It first describes the DP 16-QAM coherent receiver system and the DSP techniques used, including carrier phase estimation. It then simulates the system using Optisystem and MATLAB software and analyzes the phase noise before carrier phase estimation and Gaussian noise after by plotting average BER versus optical signal-to-noise ratio for various filter types and orders. The results show that the 3rd order Gaussian filter provided the lowest average BER and therefore the best noise performance.
The document summarizes research on energy efficient hybrid precoding for simultaneous wireless information and power transfer (SWIPT)-enabled massive MIMO non-orthogonal multiple access (NOMA) systems. It proposes a novel approach using an analog circuit-embedded (ACE) algorithm to design the analog precoder based on a switch-incorporated (SI) architecture. The ACE algorithm uses a probabilistic model and smoothed updating procedure to generate the analog precoding matrix with elements of ±1/√N. Simulation results show the proposed method achieves higher spectral and energy efficiency compared to existing techniques.
A Peak to Average Power Ratio (PAPR) Reduction in OFDM SystemsIRJET Journal
This document discusses peak-to-average power ratio (PAPR) reduction techniques for orthogonal frequency division multiplexing (OFDM) systems. It begins with an introduction to OFDM and the problem of high PAPR values in OFDM signals. It then describes the clipping and filtering method and parabolic peak cancellation method for PAPR reduction. It analyzes these techniques by evaluating complementary cumulative distribution function (CCDF) curves and bit error rate (BER) with the goal of minimizing PAPR while maintaining acceptable BER. Power amplifier nonlinearity is also discussed as a key factor affected by high PAPR OFDM signals.
This document discusses the effects of time offset (TO) and carrier frequency offset (CFO) on orthogonal frequency division multiplexing (OFDM) systems and proposes interference cancellation techniques. It introduces the OFDM system model and how CFO causes loss of orthogonality between subcarriers, resulting in inter-carrier interference (ICI) that degrades performance. The paper proposes an ICI reduction scheme using self-cancellation and evaluates its performance compared to standard OFDM. It also discusses using space-time block coding (STBC) with OFDM to improve performance by reducing bit error rate (BER) under different signal-to-noise ratios. Simulation results show STBC effectively mitigates the effects of inter-
This paper proposes a novel design for a high-speed six-transistor full adder using a two-transistor XOR gate to reduce power dissipation and area. Previous full adder designs used more transistors, resulting in higher power consumption and area. The proposed design uses a two-transistor XOR gate as a building block for an eight-transistor full adder. Simulation results show the new design has lower power consumption and transistor count compared to previous designs.
Compression: Video Compression (MPEG and others)danishrafiq
This document provides an overview of video compression techniques used in standards like MPEG and H.261. It discusses how uncompressed video data requires huge storage and bandwidth that compression aims to address. It explains that lossy compression methods are needed to achieve sufficient compression ratios. The key techniques discussed are intra-frame coding using DCT and quantization similar to JPEG, and inter-frame coding using motion estimation and compensation to remove temporal redundancy between frames. Motion vectors are found using techniques like block matching and sum of absolute differences. MPEG and other standards use a combination of these intra and inter-frame coding techniques to efficiently compress video for storage and transmission.
Simulation of ofdm modulation adapted to the transmission of a fixed imageIAEME Publication
This document summarizes a study on using OFDM modulation adapted for transmitting a fixed image over a disturbed channel. OFDM modulation was simulated using MATLAB. Both classic OFDM with a 25% guard interval and modified OFDM with a reduced guard interval below 25% were evaluated. Results were presented for various M-PSK modulation formats including BPSK, QPSK, 16PSK and 256PSK. Convolutional coding was also used to improve transmission quality. The performance of the system was evaluated in terms of the visual quality of the received image and parameters like SNR, BER under different modulation schemes and guard interval durations. Transmission of up to 98% of the original image quality was achieved.
Point Sum Average Peak Algorithm Detection of LTEAM Publications
This paper proposes an improved detection algorithm for LTE Random Access preamble detection and evaluates the performance of the algorithm with respect to the performance of algorithms proposed in the literature using MDP metric.
Cooperative partial transmit sequence for papr reduction in space frequency b...IAEME Publication
This document discusses a proposed Cooperative Partial Transmit Sequence (Co-PTS) technique for reducing Peak-to-Average Power Ratio (PAPR) in Space Frequency Block Code (SFBC) Multiple-Input Multiple-Output Orthogonal Frequency Division Multiplexing (MIMO-OFDM) signals. The proposed Co-PTS technique combines alternate optimization and spatial sub-block circular permutation. Alternate optimization reduces computational complexity while spatial sub-block circular permutation increases the number of candidate sequences, improving PAPR reduction performance. Simulation results show the proposed Co-PTS technique achieves a lower PAPR of 4.7dB compared to previous PAPR reduction techniques for MIMO-OFDM and SF
Comparative Analysis of DP QPSK and DP 16-QAM Optical Coherent Receiver, with...IRJET Journal
This document compares DP QPSK and DP 16-QAM optical coherent receivers in terms of average bit error rate (BER) when analyzing phase noise. It simulates a 112 Gbps DP 16-QAM and DP QPSK coherent receiver system with digital signal processing (DSP) using Optisystem and MATLAB. The analysis introduces noise before the receiver by varying the optical signal-to-noise ratio (OSNR) and measures average BER. Graphs of average BER versus OSNR are produced for different digital filters and filter orders to determine the filter with minimum phase noise. The DP 16-QAM system shows better power spectrum confinement and is analyzed in more detail.
This paper aims, a 3D-Pilot Aided Multi-Input Multi-Output Orthogonal Frequency Division Multiplexing (MIMO-OFDM) Channel Estimation (CE) for Digital Video Broadcasting -T2 (DVB-T2) for the 5 different proposed block and comb pilot patterns model and performed on different antenna configuration. The effects of multi-transceiver antenna on channel estimation are addressed with different pilot position in frequency, time and the vertical direction of spatial domain framing. This paper first focus on designing of 5-different proposed spatial correlated pilot pattern model with optimization of pilot overhead. Then it demonstrates the performance comparison of Least Square (LS) & Linear Minimum Mean Square Error (LMMSE), two linear channel estimators for 3D-Pilot Aided patterns on different antenna configurations in terms of Bit Error Rate. The simulation results are shown for Rayleigh fading noise channel environments. Also, 3x4 MIMO configuration is recommended as the most suitable configuration in this noise channel environments.
An Advanced Implementation of a Digital Artificial Reverberatora3labdsp
This paper proposes an enhanced hybrid reverberator that uses both measured impulse responses and synthesized impulse responses to model reverberation effects. It develops an automatic procedure to set the parameters of the hybrid reverberator by analyzing the mixing time of measured impulse responses and minimizing a loss function in the cepstral domain. Experimental results show the proposed method produces higher quality reverberation effects than a previous method, with only a small increase in computational cost, as confirmed by listening tests.
A NOVEL APPROACH TO CHANNEL DECORRELATION FOR STEREO ACOUSTIC ECHO CANCELLATI...a3labdsp
This document proposes a novel approach to decorrelating stereo acoustic signals for acoustic echo cancellation based on the psychoacoustic phenomenon of the "missing fundamental". The approach tracks and removes the pitch from one channel of the stereo signal using an adaptive notch filter, which greatly reduces inter-channel coherence in the lower spectrum without affecting signal quality. Experimental results show the proposed approach provides significant coherence reduction and faster convergence speed of adaptive filters compared to a masked noise injection method, while better preserving the stereo quality.
Low Power High-Performance Computing on the BeagleBoard Platforma3labdsp
The ever increasing energy requirements of supercomputers and server farms is driving the scientific and industrial communities to take in deeper consideration the energy efficiency of computing equipments. This contribution addresses the issue proposing a cluster of ARM processors for high-performance computing. The cluster is composed of five BeagleBoard-xM, with one board managing the cluster, and the other boards executing the actual processing. The software platform is based on the Angstrom GNU/Linux distribution and is equipped with a distributed file system to ease sharing data and code among the nodes of the cluster, and with tools for managing tasks and monitoring the status of each node. The computational capabilities of the cluster have been assessed through High-Performance Linpack and a cluster-wide speaker diarization algorithm, while power consumption has been measured using a clamp meter. Experimental results obtained in the speaker diarization task showed that the energy efficiency of the BeagleBoard-xM cluster is comparable to the one of a laptop computer equipped with a Intel Core2 Duo T8300 running at 2.4 GHz. Furthermore, removing the bottleneck due to the Ethernet interface, the BeagleBoard-xM cluster is able to achieve a superior energy efficiency.
Hybrid Reverberator Using Multiple Impulse Responses for Audio Rendering Impr...a3labdsp
The document proposes an algorithm for audio rendering using multiple impulse responses that allows for reproduction of a moving listener position. It analyzes impulse response tails to generate a prototype tail and uses a hybrid reverberation structure including FIR and IIR filters to synthesize the reverberation effect in real-time. Experimental results on a church impulse response database show the approach can accurately reproduce reverberation time and clarity measurements compared to real impulse responses. Informal listening tests found no perceptible differences between the proposed approach and an existing technique.
A Low Latency Implementation of a Non Uniform Partitioned Overlap and Save Al...a3labdsp
FIR convolution is a widely used operation in digital signal processing field, especially for filtering operations in real time scenarios. In this context, low computationally demanding techniques for calculating convolutions with low input/output latency become essential, considering that the real time requirements are strictly related to the impulse response length. In this paper, a multithreading real time implementation of a Non Uniform Partitioned Overlap and Save algorithm is proposed with the aim of lowering the workload required in applications like reverberation, also exploiting the human ear sensitivity. Several results are reported in order to show the effectiveness of the proposed approach in terms of computational cost, taking into consideration different impulse responses and also introducing comparisons with existing techniques of the state of the art.
System Identification Based on Hammerstein Models Using Cubic Splinesa3labdsp
The document presents a new approach for identifying Hammerstein models, which are nonlinear systems composed of a static nonlinearity followed by a linear filter. The approach uses an adaptive Catmull-Rom cubic spline to model the static nonlinearity, instead of high-order polynomials. Experimental results on simulated and real-world systems show the spline-based approach more accurately identifies the nonlinear characteristics and outperforms an existing polynomial-based technique, especially for highly nonlinear systems. The linear filter is modeled using an adaptive IIR filter.
Approximation of Real Impulse Response Using IIR Structures a3labdsp
In this paper, we propose a new approach to the approximation and simulation of a real impulse response. Starting from a preliminary analysis of the mixing time, the impulse response is decomposed in the time domain considering the early and late reflections. Therefore, an IIR structure composed of a cascade of second-order sections and four all-pass filters is employed to synthesize the first part of the impulse response, using a parametric optimization process in the frequency domain. Then, a recursive structure composed of comb and all-pass filters is used to synthesize the late reflections, exploiting a minimization criterion in the cepstral domain. Several results are reported taking into consideration a real impulse response, confirming the validity of the proposed approach.
A Distributed System for Recognizing Home Automation Commands and Distress Ca...a3labdsp
The document describes a distributed system that recognizes home automation commands and distress calls in Italian. It consists of two units: a Local Multimedia Control Unit that recognizes commands/calls and manages communication, and a Central Management Unit that integrates home services and handles emergencies. The system uses acoustic echo cancellation and speech recognition to understand commands even in noisy environments. An evaluation of the system showed it achieved over 90% accuracy on headset microphone data and over 50% on distant microphone data.
International Journal of Engineering Research and Development (IJERD)IJERD Editor
journal publishing, how to publish research paper, Call For research paper, international journal, publishing a paper, IJERD, journal of science and technology, how to get a research paper published, publishing a paper, publishing of journal, publishing of research paper, reserach and review articles, IJERD Journal, How to publish your research paper, publish research paper, open access engineering journal, Engineering journal, Mathemetics journal, Physics journal, Chemistry journal, Computer Engineering, Computer Science journal, how to submit your paper, peer reviw journal, indexed journal, reserach and review articles, engineering journal, www.ijerd.com, research journals,
yahoo journals, bing journals, International Journal of Engineering Research and Development, google journals, hard copy of journal
International Journal of Engineering Research and Development (IJERD)IJERD Editor
This document summarizes a research paper on hardware efficient reconfigurable FIR filters. It discusses two new architectures proposed: the constant shifts method (CSM) and programmable shifts method (PSM). CSM partitions coefficients into fixed groups and stores them directly in a lookup table. PSM eliminates redundancy in coefficients using a binary common subexpression algorithm before storing in a coded format. Both methods use a shift-and-add unit and multiplexers to efficiently implement coefficient multiplication and allow reconfiguration for different standards. The architectures aim to integrate reconfigurability with low complexity for FIR filters used in wireless communications.
Iaetsd gmsk modulation implementation for gsm in dspIaetsd Iaetsd
This document describes the implementation of a GMSK modulator on a TMS320C6713 digital signal processor. GMSK modulation is used in GSM cellular systems due to its bandwidth efficiency. The author designed a simple algorithm to accurately generate GMSK signals in DSP. Key components included a numerically controlled oscillator and Gaussian low-pass filter implemented as a finite impulse response filter. Simulation results were obtained using Elanix software to verify the GMSK modulator design.
Performance Analysis of Acoustic Echo Cancellation TechniquesIJERA Editor
Mainly, the adaptive filters are implemented in time domain which works efficiently in most of the applications. But in many applications the impulse response becomes too large, which increases the complexity of the adaptive filter beyond a level where it can no longer be implemented efficiently in time domain. An example of where this can happen would be acoustic echo cancellation (AEC) applications. So, there exists an alternative solution i.e. to implement the filters in frequency domain. AEC has so many applications in wide variety of problems in industrial operations, manufacturing and consumer products. Here in this paper, a comparative analysis of different acoustic echo cancellation techniques i.e. Frequency domain adaptive filter (FDAF), Least mean square (LMS), Normalized least mean square (NLMS) &Sign error (SE) is presented. The results are compared with different values of step sizes and the performance of these techniques is measured in terms of Error rate loss enhancement (ERLE), Mean square error (MSE)& Peak signal to noise ratio (PSNR).
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.
An fpga implementation of the lms adaptive filter eSAT Journals
This document describes an FPGA implementation of the Least Mean Square (LMS) adaptive filter algorithm for active vibration control. It compares fixed-point and floating-point implementations in terms of area usage and performance. The LMS algorithm is implemented using a finite state machine model with separate modules for operations like filtering, error estimation, and weight adaptation. Both implementations utilize this structural model. The fixed-point version uses 16-bit integers and fractions, while the floating-point version leverages IP cores. Results show the floating-point implementation has better accuracy and resource utilization than the fixed-point version for active vibration control applications on FPGAs.
Design and Implementation of an Embedded System for Software Defined RadioIJECEIAES
In this paper, developing high performance software for demanding real-time embed- ded systems is proposed. This software-based design will enable the software engineers and system architects in emerging technology areas like 5G Wireless and Software Defined Networking (SDN) to build their algorithms. An ADSP-21364 floating point SHARC Digital Signal Processor (DSP) running at 333 MHz is adopted as a platform for an embedded system. To evaluate the proposed embedded system, an implementation of frame, symbol and carrier phase synchronization is presented as an application. Its performance is investigated with an on line Quadrature Phase Shift keying (QPSK) receiver. Obtained results show that the designed software is implemented successfully based on the SHARC DSP which can utilized efficiently for such algorithms. In addition, it is proven that the proposed embedded system is pragmatic and capable of dealing with the memory constraints and critical time issue due to a long length interleaved coded data utilized for channel coding.
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.
This document summarizes a technique called mixed single frequency delay line filtering that is proposed to optimize computational complexity and processing delay for multichannel audio crosstalk cancellation. The technique combines mixed filtering and single frequency delay line filtering. Mixed filtering allows all filtering operations to be performed in a single equation in the frequency domain, reducing computations. Single frequency delay line filtering partitions long filter impulse responses into shorter partitions that are filtered using overlap-save, reducing delay. The proposed technique partitions impulse responses and applies mixed filtering to further reduce computations over existing methods. It is shown to provide less computational complexity and delay than overlap-save filtering while maintaining performance for long impulse responses.
Design and optimization of a new compact 2.4 GHz-bandpass filter using DGS te...TELKOMNIKA JOURNAL
The objective of this work is the study, the design and the optimization of an innovative structure of a network of coupled copper metal lines deposited on the upper surface of a R04003 type substrate of height 0.813 with a ground deformed by slots (DGS). This structure is designed in an optimal configuration for use in the design of narrowband bandpass filter for wireless communication systems (WLAN), the aim of use the defected ground structure is to remove the unwanted harmonics in the rejection band, the simulation results obtained from this structure using CST software show a very high selectivity of the designed filter, a very low level of losses (less than-0.45 dB) with a size overall size of 43.5x34.3 mm.
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.
International Journal of Engineering Research and DevelopmentIJERD Editor
This document describes a proposed VLSI architecture for an optimized low power digit serial finite impulse response (FIR) filter using multiple constant multiplications (MCM). It introduces an algorithm to optimize the area of digit serial MCM operations at the gate level by considering implementation costs of digit serial addition, subtraction, and shift operations. The proposed filter architecture aims to reduce area and power compared to designs using generic digit serial multipliers through the use of MCM blocks optimized for area. Experimental results indicate the algorithm leads to lower complexity digit serial MCM designs.
This document summarizes a research paper on designing low power digit serial finite impulse response (FIR) filters using multiple constant multiplication (MCM) techniques. It proposes an architecture that optimizes the area of digit serial MCM operations at the gate level by considering the implementation costs of digit serial addition, subtraction and shift operations. An algorithm is presented to design digit serial FIR filters under a shift-adds architecture to reduce area compared to designs using generic digit serial multipliers. Experimental results show the technique leads to lower complexity digit serial MCM designs.
Types Of Window Being Used For The Selected GranuleLeslie Lee
The document discusses different types of mode selective devices that can be used for mode multiplexing over few-mode fiber. Free-space based devices are bulky while fiber based devices are more compact and easier to integrate. Early demonstrations transmitted data over 107 Gb/s using the LP01 and LP11 fiber modes and 58.8 Gb/s using dual modes with electronic MIMO processing for mode separation. Mode selective devices can be categorized as either free-space based or fiber based, with fiber based being preferable due to their compact size and integration capabilities.
First order sigma delta modulator with low-power consumption implemented in a...eSAT Journals
Abstract
This paper presents a design of a switched-capacitor discrete time 1st order Delta-Sigma modulator used for a resolution of 8 bits
Sigma-Delta analog to digital converter. For lower power consumption, the use of operational transconductance amplifier is
necessary in order to provide wide output voltage swing and moderate DC gain. Simulation results showed that with 0.35um CMOS
technology, 80 KHz signal bandwidth and oversampling rate of 64, the modulator achieved 49.25 dB Signal to Noise Ratio (SNR) and
the power consumption was 5.5 mW under ±1.5V supply voltage .
Index terms: Analog-to-Digital conversion, Delta-Sigma modulation, CMOS technology, Transconductance operational
amplifier.
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.
International Journal of Engineering Inventions (IJEI) provides a multidisciplinary passage for researchers, managers, professionals, practitioners and students around the globe to publish high quality, peer-reviewed articles on all theoretical and empirical aspects of Engineering and Science.
The peer-reviewed International Journal of Engineering Inventions (IJEI) is started with a mission to encourage contribution to research in Science and Technology. Encourage and motivate researchers in challenging areas of Sciences and Technology.
The document compares the performance of a Root Raised Cosine matched filter implemented using hybrid-logarithmic arithmetic versus standard binary and floating point arithmetic. Simulations showed that the hybrid logarithmic structure offered superior performance to fixed point solutions while having significantly reduced complexity compared to floating point equivalents. The use of hybrid logarithmic arithmetic also has the potential to reduce power consumption, latency, and hardware complexity for mobile applications.
A blind channel shortening for multiuser, multicarrier CDMA system over multi...TELKOMNIKA JOURNAL
In this paper we derive the Multicarrier Equalization by Restoration of Redundancy (MERRY) algorithm: A blind, adaptive channel shortening algorithm for updating a Time-domain Equalizer (TEQ) in a system employing MultiCarrier Code Division Multiple Access (MC-CDMA) modulation. We show that the MERRY algorithm applied to the MC-CDMA system converges considerably more rapidly than in the Orthogonal Frequency Division Multiplexing (OFDM) system [1]. Simulations results are provided to demonstrate the performance of the algorithm.
IRJET- Efficient Shift add Implementation of Fir Filter using Variable Pa...IRJET Journal
This document discusses efficient implementations of shift-add operations in finite impulse response (FIR) filters using variable partition hybrid form structures. FIR filters are widely used in digital signal processing and their performance is dominated by multiplication operations. The proposed method aims to reduce power consumption and complexity by implementing multiplications using optimized shift-add networks instead of multipliers. It explores variable size partitioning approaches and prefix adders to reduce gate count, dynamic power, and improve filter performance.
Similar to Optimized implementation of an innovative digital audio equalizer (20)
Evaluation of a Multipoint Equalization System based on Impulse Responses Pro...a3labdsp
This document evaluates a multipoint equalization algorithm that combines fractional octave smoothing of impulse responses measured in multiple locations in a room or car. It investigates how the equalization performance is affected by varying parameters like the number of measurement positions and equalization zone size. The algorithm extracts a representative prototype response and inverse filter from the smoothed impulse responses. Tests show the proposed approach achieves better spectral deviation and equalization than a single-point method, and performance decreases but remains effective as the equalization zone is expanded to more distant positions.
Hybrid Reverberation Algorithm: a Practical Approacha3labdsp
Reverberation is a well known eect that has an important role in our listening experience. Reverberation changes positively the perception of the sound, adding fullness and sense of space. Generally, two approaches are employed for articial reverberation: the desired signal can be obtained by convolving the input signal
with a measured impulse response (IR) or by synthetic techniques based on recursive lter structures. Taking into account the advantages of both approaches, a hybrid articial reverberation algorithm is presented aiming to reproduce the acoustic behaviour of real environment with a low computational load. More in detail, the early reflections are derived from a real impulse response, truncated considering the calculated mixing time, and the reverberation tail is obtained using an IIR lter network. The parameters dening this structure are automatically derived from the analyzed impulse response, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The effectiveness of the proposed approach has been proved taking into account a real Italian Theatre impulse response providing comparison with the existing state-of-art techniques in terms of objective and subjective measures.
Mixed Time Frequency Approach for Multipoint Room Response Equalizationa3labdsp
A still open problem in the field of room response equalization is the development of perceptually useful mixed-phase equalizers. In a recent paper, a multipoint mixed-phase room response equalization system, integrating a minimum-phase multiple position room magnitude equalizer and a FIR group delay equalizer, was developed in the frequency domain. Starting from this approach, a mixed time-frequency algorithm is here proposed. The minimum-phase multiple position equalizer developed in the frequency domain, is combined with an all-pass FIR phase equalizer, designed in the time domain considering a suitable time-reversed version of a prototype function and taking advantage of the mixing time evaluation. Several tests have been performed considering real environments and comparing the proposed approach with the previous one, based on a group delay compensation. Subjective listening tests have also been done in a real environment, confirming the improvement in the perceived audio quality.
Audio Morphing for Percussive Sound Generationa3labdsp
The aim of audio morphing algorithms is to combine two or more sounds to create a new sound with intermediate timbre and duration. During the last two decades several efforts have been made to improve morphing algorithms in order to obtain more realistic and perceptually relevant sounds. In this paper we present an automatic audio morphing technique applied to percussive musical instruments. Based on preprocessing of the sound references in frequency domain and linear interpolation in time domain, the presented approach allows one to generate high quality hybrid sounds at a low computational cost. Several results are reported in order to show the effectiveness of the proposed approach in terms of audio quality and acoustic perception of the generated hybrid sounds, taking into consideration different percussive samples. Mean opinion score and multidimensional scaling were used to compare the presented approach with existing state of the art techniques.
An Efficient DSP Implementation of a Dynamic Convolution Using Principal Comp...a3labdsp
In the recent years, several techniques have been proposed in the literature in order to attempt the emulation of nonlinear electro-acoustic devices, such as compressors, limiters, and pre-amplifiers. Among them, the dynamic convolution technique is one of the most common approaches used to perform a nonlinear convolution. In this paper, an efficient DSP implementation of a nonlinear system emulation based on the dynamic convolution technique and principal component analysis is proposed with the aim of lowering the required workload and the global memory usage. Several results are reported in order to show the effectiveness of the proposed approach, taking into consideration a guitar pre-amplifier as a particular case study and also introducing comparisons with the existing techniques of the state of the art.
Approximation of Dynamic Convolution Exploiting Principal Component Analysis:...a3labdsp
In recent years, several techniques have been proposed in the literature in order to attempt the emulation of nonlinear electro-acoustic devices, such as compressors, distortions, and preamplifiers. Among them, the dynamic convolution technique is one of the most common approaches used to perform this task. In this paper an exhaustive objective and subjective analysis of a dynamic convolution operation based on principal components analysis has been performed. Taking into consideration real nonlinear systems, such as bass preamplifier, distortion, and compressor, comparisons with the existing techniques of the state of the art have been carried out in order to prove the effectiveness of the proposed approach.
An Efficient DSP Based Implementation of a Fast Convolution Approach with non...a3labdsp
"Finite impulse response convolution is one of the most widely used operation in digital signal processing field for filtering operations. In this context, low computationally demanding techniques become essential for calculating convolutions with low input/output latency in real scenarios, considering that the real time requirements are strictly related to the impulse response length. In this context, an efficient DSP implementation of a fast convolution approach is presented with the aim of lowering the workload required in applications like reverberation. It is based on a non uniform partitioning of the impulse response and a psychoacoustic technique derived from the human ear sensitivity. Several results are reported in order to prove the effectiveness of the proposed approach also introducing comparisons with the existing techniques of the state of the art."
A Hybrid Approach for Real-time Room Acoustic Response Simulationa3labdsp
Reverberation is a well known effect particularly important for music listening especially for recorded and live music. Generally, there are two approaches for artificial reverberation: the desired signal can be obtained by convolving the input signal with a measured impulse response (IR) or a synthetic one. Taking into account the advantages of both approaches, a hybrid artificial reverberation algorithm is presented. The early reflections are derived from a real IR, truncated considering the calculated mixing time, and the reverberation tail is obtained considering the Moorer's structure. The parameters defining this structure are derived from the analyzed IR, using a minimization criteria based on Simultaneous Perturbation Stochastic Approximation (SPSA). The obtained results showed a high-quality reverberator with a low computational load.
Salesforce Integration for Bonterra Impact Management (fka Social Solutions A...Jeffrey Haguewood
Sidekick Solutions uses Bonterra Impact Management (fka Social Solutions Apricot) and automation solutions to integrate data for business workflows.
We believe integration and automation are essential to user experience and the promise of efficient work through technology. Automation is the critical ingredient to realizing that full vision. We develop integration products and services for Bonterra Case Management software to support the deployment of automations for a variety of use cases.
This video focuses on integration of Salesforce with Bonterra Impact Management.
Interested in deploying an integration with Salesforce for Bonterra Impact Management? Contact us at sales@sidekicksolutionsllc.com to discuss next steps.
Ivanti’s Patch Tuesday breakdown goes beyond patching your applications and brings you the intelligence and guidance needed to prioritize where to focus your attention first. Catch early analysis on our Ivanti blog, then join industry expert Chris Goettl for the Patch Tuesday Webinar Event. There we’ll do a deep dive into each of the bulletins and give guidance on the risks associated with the newly-identified vulnerabilities.
Building Production Ready Search Pipelines with Spark and MilvusZilliz
Spark is the widely used ETL tool for processing, indexing and ingesting data to serving stack for search. Milvus is the production-ready open-source vector database. In this talk we will show how to use Spark to process unstructured data to extract vector representations, and push the vectors to Milvus vector database for search serving.
zkStudyClub - LatticeFold: A Lattice-based Folding Scheme and its Application...Alex Pruden
Folding is a recent technique for building efficient recursive SNARKs. Several elegant folding protocols have been proposed, such as Nova, Supernova, Hypernova, Protostar, and others. However, all of them rely on an additively homomorphic commitment scheme based on discrete log, and are therefore not post-quantum secure. In this work we present LatticeFold, the first lattice-based folding protocol based on the Module SIS problem. This folding protocol naturally leads to an efficient recursive lattice-based SNARK and an efficient PCD scheme. LatticeFold supports folding low-degree relations, such as R1CS, as well as high-degree relations, such as CCS. The key challenge is to construct a secure folding protocol that works with the Ajtai commitment scheme. The difficulty, is ensuring that extracted witnesses are low norm through many rounds of folding. We present a novel technique using the sumcheck protocol to ensure that extracted witnesses are always low norm no matter how many rounds of folding are used. Our evaluation of the final proof system suggests that it is as performant as Hypernova, while providing post-quantum security.
Paper Link: https://eprint.iacr.org/2024/257
For the full video of this presentation, please visit: https://www.edge-ai-vision.com/2024/06/temporal-event-neural-networks-a-more-efficient-alternative-to-the-transformer-a-presentation-from-brainchip/
Chris Jones, Director of Product Management at BrainChip , presents the “Temporal Event Neural Networks: A More Efficient Alternative to the Transformer” tutorial at the May 2024 Embedded Vision Summit.
The expansion of AI services necessitates enhanced computational capabilities on edge devices. Temporal Event Neural Networks (TENNs), developed by BrainChip, represent a novel and highly efficient state-space network. TENNs demonstrate exceptional proficiency in handling multi-dimensional streaming data, facilitating advancements in object detection, action recognition, speech enhancement and language model/sequence generation. Through the utilization of polynomial-based continuous convolutions, TENNs streamline models, expedite training processes and significantly diminish memory requirements, achieving notable reductions of up to 50x in parameters and 5,000x in energy consumption compared to prevailing methodologies like transformers.
Integration with BrainChip’s Akida neuromorphic hardware IP further enhances TENNs’ capabilities, enabling the realization of highly capable, portable and passively cooled edge devices. This presentation delves into the technical innovations underlying TENNs, presents real-world benchmarks, and elucidates how this cutting-edge approach is positioned to revolutionize edge AI across diverse applications.
Digital Marketing Trends in 2024 | Guide for Staying AheadWask
https://www.wask.co/ebooks/digital-marketing-trends-in-2024
Feeling lost in the digital marketing whirlwind of 2024? Technology is changing, consumer habits are evolving, and staying ahead of the curve feels like a never-ending pursuit. This e-book is your compass. Dive into actionable insights to handle the complexities of modern marketing. From hyper-personalization to the power of user-generated content, learn how to build long-term relationships with your audience and unlock the secrets to success in the ever-shifting digital landscape.
HCL Notes and Domino License Cost Reduction in the World of DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-and-domino-license-cost-reduction-in-the-world-of-dlau/
The introduction of DLAU and the CCB & CCX licensing model caused quite a stir in the HCL community. As a Notes and Domino customer, you may have faced challenges with unexpected user counts and license costs. You probably have questions on how this new licensing approach works and how to benefit from it. Most importantly, you likely have budget constraints and want to save money where possible. Don’t worry, we can help with all of this!
We’ll show you how to fix common misconfigurations that cause higher-than-expected user counts, and how to identify accounts which you can deactivate to save money. There are also frequent patterns that can cause unnecessary cost, like using a person document instead of a mail-in for shared mailboxes. We’ll provide examples and solutions for those as well. And naturally we’ll explain the new licensing model.
Join HCL Ambassador Marc Thomas in this webinar with a special guest appearance from Franz Walder. It will give you the tools and know-how to stay on top of what is going on with Domino licensing. You will be able lower your cost through an optimized configuration and keep it low going forward.
These topics will be covered
- Reducing license cost by finding and fixing misconfigurations and superfluous accounts
- How do CCB and CCX licenses really work?
- Understanding the DLAU tool and how to best utilize it
- Tips for common problem areas, like team mailboxes, functional/test users, etc
- Practical examples and best practices to implement right away
Let's Integrate MuleSoft RPA, COMPOSER, APM with AWS IDP along with Slackshyamraj55
Discover the seamless integration of RPA (Robotic Process Automation), COMPOSER, and APM with AWS IDP enhanced with Slack notifications. Explore how these technologies converge to streamline workflows, optimize performance, and ensure secure access, all while leveraging the power of AWS IDP and real-time communication via Slack notifications.
In the realm of cybersecurity, offensive security practices act as a critical shield. By simulating real-world attacks in a controlled environment, these techniques expose vulnerabilities before malicious actors can exploit them. This proactive approach allows manufacturers to identify and fix weaknesses, significantly enhancing system security.
This presentation delves into the development of a system designed to mimic Galileo's Open Service signal using software-defined radio (SDR) technology. We'll begin with a foundational overview of both Global Navigation Satellite Systems (GNSS) and the intricacies of digital signal processing.
The presentation culminates in a live demonstration. We'll showcase the manipulation of Galileo's Open Service pilot signal, simulating an attack on various software and hardware systems. This practical demonstration serves to highlight the potential consequences of unaddressed vulnerabilities, emphasizing the importance of offensive security practices in safeguarding critical infrastructure.
Trusted Execution Environment for Decentralized Process MiningLucaBarbaro3
Presentation of the paper "Trusted Execution Environment for Decentralized Process Mining" given during the CAiSE 2024 Conference in Cyprus on June 7, 2024.
GraphRAG for Life Science to increase LLM accuracyTomaz Bratanic
GraphRAG for life science domain, where you retriever information from biomedical knowledge graphs using LLMs to increase the accuracy and performance of generated answers
HCL Notes und Domino Lizenzkostenreduzierung in der Welt von DLAUpanagenda
Webinar Recording: https://www.panagenda.com/webinars/hcl-notes-und-domino-lizenzkostenreduzierung-in-der-welt-von-dlau/
DLAU und die Lizenzen nach dem CCB- und CCX-Modell sind für viele in der HCL-Community seit letztem Jahr ein heißes Thema. Als Notes- oder Domino-Kunde haben Sie vielleicht mit unerwartet hohen Benutzerzahlen und Lizenzgebühren zu kämpfen. Sie fragen sich vielleicht, wie diese neue Art der Lizenzierung funktioniert und welchen Nutzen sie Ihnen bringt. Vor allem wollen Sie sicherlich Ihr Budget einhalten und Kosten sparen, wo immer möglich. Das verstehen wir und wir möchten Ihnen dabei helfen!
Wir erklären Ihnen, wie Sie häufige Konfigurationsprobleme lösen können, die dazu führen können, dass mehr Benutzer gezählt werden als nötig, und wie Sie überflüssige oder ungenutzte Konten identifizieren und entfernen können, um Geld zu sparen. Es gibt auch einige Ansätze, die zu unnötigen Ausgaben führen können, z. B. wenn ein Personendokument anstelle eines Mail-Ins für geteilte Mailboxen verwendet wird. Wir zeigen Ihnen solche Fälle und deren Lösungen. Und natürlich erklären wir Ihnen das neue Lizenzmodell.
Nehmen Sie an diesem Webinar teil, bei dem HCL-Ambassador Marc Thomas und Gastredner Franz Walder Ihnen diese neue Welt näherbringen. Es vermittelt Ihnen die Tools und das Know-how, um den Überblick zu bewahren. Sie werden in der Lage sein, Ihre Kosten durch eine optimierte Domino-Konfiguration zu reduzieren und auch in Zukunft gering zu halten.
Diese Themen werden behandelt
- Reduzierung der Lizenzkosten durch Auffinden und Beheben von Fehlkonfigurationen und überflüssigen Konten
- Wie funktionieren CCB- und CCX-Lizenzen wirklich?
- Verstehen des DLAU-Tools und wie man es am besten nutzt
- Tipps für häufige Problembereiche, wie z. B. Team-Postfächer, Funktions-/Testbenutzer usw.
- Praxisbeispiele und Best Practices zum sofortigen Umsetzen
5th LF Energy Power Grid Model Meet-up SlidesDanBrown980551
5th Power Grid Model Meet-up
It is with great pleasure that we extend to you an invitation to the 5th Power Grid Model Meet-up, scheduled for 6th June 2024. This event will adopt a hybrid format, allowing participants to join us either through an online Mircosoft Teams session or in person at TU/e located at Den Dolech 2, Eindhoven, Netherlands. The meet-up will be hosted by Eindhoven University of Technology (TU/e), a research university specializing in engineering science & technology.
Power Grid Model
The global energy transition is placing new and unprecedented demands on Distribution System Operators (DSOs). Alongside upgrades to grid capacity, processes such as digitization, capacity optimization, and congestion management are becoming vital for delivering reliable services.
Power Grid Model is an open source project from Linux Foundation Energy and provides a calculation engine that is increasingly essential for DSOs. It offers a standards-based foundation enabling real-time power systems analysis, simulations of electrical power grids, and sophisticated what-if analysis. In addition, it enables in-depth studies and analysis of the electrical power grid’s behavior and performance. This comprehensive model incorporates essential factors such as power generation capacity, electrical losses, voltage levels, power flows, and system stability.
Power Grid Model is currently being applied in a wide variety of use cases, including grid planning, expansion, reliability, and congestion studies. It can also help in analyzing the impact of renewable energy integration, assessing the effects of disturbances or faults, and developing strategies for grid control and optimization.
What to expect
For the upcoming meetup we are organizing, we have an exciting lineup of activities planned:
-Insightful presentations covering two practical applications of the Power Grid Model.
-An update on the latest advancements in Power Grid -Model technology during the first and second quarters of 2024.
-An interactive brainstorming session to discuss and propose new feature requests.
-An opportunity to connect with fellow Power Grid Model enthusiasts and users.
Optimized implementation of an innovative digital audio equalizer
1. ID 86
Optimized Implementation of an
Innovative Digital Audio Equalizer
Marco Virgulti1
Stefania Cecchi1
Andrea Primavera1
Laura Romoli1
Francesco Piazza1
Ferruccio Bettarelli2
Emanuele Ciavattini2
1A3LAB, DII, Universit´a Politecnica delle Marche,Via Brecce Bianche, 60131 Ancona, Italy
2Leaff Engineering, Via Pastore 10, 60027 Ancona, Italy
Correspondence should be addressed to Stefania Cecchi (s.cecchi@univpm.it)
Abstract
Digital audio equalization is one of the most common operations in the acoustic field, but its perfor-
mance 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 com-
paring 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.
2. Introduction
Digital audio equalization is one of the most common operations in the acoustic field, but its performance
is strictly related to the adopted filter design techniques.
The equalization purpose is to enhance the listening experience, preserving a linear phase response
with the lowest delay and the lowest computational complexity.
In order to have linear phase response, FIR filters are usually employed both in time and frequency
domain.
State of the art
An efficient digital equalizer can be implemented using:
a tree structured filter bank[1]: the analysis filter bank was built with equal stages splitting the input
signal in two subbands while the synthesis filter bank has to recombine back the bands.☛
✡
✟
✠Drawbacks: too high delay that exponentially increases with the number of subbands
a frequency masking technique[2]; computationally efficient techniques for the design of sharp
low-pass, high-pass, band-pass, and band-stop filters with arbitrary passband.☛
✡
✟
✠Drawbacks: the introduced delay dramatically increases with stricter constraints.
Remez algorithm[3]: when the response of adjacent bands are added together, if the composite
frequency response shows an unacceptable error deviations, a new filter with a new stopband cutoff
frequency has to be designed.☛
✡
✟
✠Drawbacks: This procedure is iterated until the deviation becomes acceptable resulting in a too high
computational complexity.
a frequency domain algorithm[4]: the equalization consists of a complex multiplication of the input
spectrum with the frequency equalization function that, when transformed in time domain, has all the
properties of a FIR filter. The computational complexity is very low.☛
✡
✟
✠Drawbacks: the algorithm efficiency is strictly bound to frequency resolution (e.g,large ripples at
bands edges are easily observed).
3. Introduction
The main disadvantages of these approaches are the too high delay and the ripples in the frequency
response, when adjacent bands are added.
A possible solution could arise from the multirate techniques applied to adaptive systems.
In this context, the problem of aliasing cancellation when an adaptive filtering is included in a
filterbank with perfect reconstruction is well-known.
Different approaches were presented such as cross terms between subbands [5] or extra terms taking
into account adjacent bands [6].
Starting from these approaches, an innovative digital audio equalizer has been introduced in [7, 8].
Taking into consideration multirate systems and their property, the idea was to realize a linear phase
FIR equalizer that overcame the well-known drawbacks.
Therefore, an optimized version of this algorithm is here proposed:
it is based on the use of IIR filters capable to reduce the required computational complexity preserving
the audio quality [9, 10].
the overall scheme of the algorithm derives from two analysis and two synthesis cosine modulated
filter banks properly combined in order to have flat response when all bands are added together,
exploiting multirate properties (as for the FIR approach).
The new technique preserve all the characteristics of the previous implementation overcoming all the
well-known problems documented in the literature, furthermore reducing the computational complexity
and improving the filters selectiveness.
4. Proposed Algorithms
Main idea
The main idea of the proposed approach is to realize an innovative equalizer using a particular filter
bank structure, capable of reducing ripples in the frequency response, when adjacent bands are added
together.
is derived from the subband adaptive filtering structure presented in [11, 6],
a double analysis/synthesis filter banks is combined employing multirate properties
starting from this, it is possible to obtain an innovative equalizer having all the advantages of this
particular solution [8, 7].
The impulse responses of the analysis/synthesis filters are the following,
hk(n) = 2h0(n)cos(
π
M
(k + 0.5)(n −
N
2
) + θk)
fk(n) = 2h0(n)cos(
π
M
(k + 0.5)(n −
N
2
) − θk)
(1)
where θk = −(1)k π
4 , k is the subband index defined between 0 and M −1, M is the number of subband,
and N is the order of the prototype filter h0(n).
Using this particular structure,
it is possible to significantly reduce the ripple amounts difference in the transition band between
adjacent filters[8, 7];
it is easily extended to higher number of bands.
increasing the number of subbands, the FIR length has to be increased in order to have good
performance at low frequency bands.
For this reason, a new solution has been proposed taking advantage of IIR filters.
5. FIR based equalizer
The FIR Equalizer has been designed
using Eq.(1)
developing h0(n) has a FIR filter prototype.
A near perfect reconstruction proto-
type realized by Kaiser Windows method
[12], with very low computational cost.
This technique modifies the 6-dB cut-off frequency
of the filter in order to obtain the 3-dB cut-off fre-
quency placed approximately at π/2M.
The function minimized is the following:
ξ = ||H(ejπ/2M)| −
1
√
2
|. (2)
H is the frequency response of the following filter
h[n] =
sin((n − N/2)ωc,6dB)
π(n − N/2)
w[n], (3)
where w[n] is a Kaiser window and ωc,6−dB is the
cut-off frequency.
Figure 1: Overall structure of the proposed equalizer
6. IIR-based equalizer
The IIR Equalizer has been designed
using Eq.(1)
developing h0(n) has a IIR filter prototype.
Also in this case, a near perfect reconstruction pro-
totype has been considered[9, 10].
The IIR prototype filter is defined as
H(z) =
2M−1
k=0
z−kPk(z2M), (4)
where Pk(z) is the k-th type-I polyphase compo-
nents of H(z), assuming that
Pk(z) =
Nk(z)
D(z)
(5)
for k = 0, 1, · · · , 2M − 1.
The polyphase component can be obtained using
the two channel cascaded lattice structure [10] as
shown in Fig.2.
The following structure is obtained by replacing z−1
of the classical FIR lattice structure with a first
order all-pass filter [9]
ck,m
-ck,m
sk,m
sk,m
+
+Am(Z)
P(m-1)
(z)k
P(m-1)
(z)M+k
P(m)
(z)k
P(m)
(z)M+k
Figure 2: IIR lattice structure for design of Pk(z)
The relation between each section is given by the
following equations:
Pm
k (z) = ck,mPm−1
k
(z) + sk,mAm(z)Pm−1
M+k
(z)
Pm
M+k(z) = sk,mPm−1
k
(z) − ck,mAm(z)Pm−1
M+k
(z)
(6)
where ck,m = cos(θk,m) and sk,m = sin(θk,m).
7. IIR-based equalizer
To satisfy the condition that all the polyphase com-
ponents have the same denominator as shown in
Eq.(5), the same all-pass filters must be used in all
the lattice structures.
It is defined as follows:
Am(z) =
am + z−1
1 + amz−1
, (7)
for m ≥ 1, k = 0, 1, · · · , M/2 − 1 and considering
|ai| < 1 for i = 1, 2, · · · , m to ensure the filter stabil-
ity.
Then the optimization is performed considering the
minimization of the stop-band energy using the fol-
lowing objective function:
φ =
π
π/(2M)+δ
H(ejω)
2
dω (8)
where δ < π/(2M).
The final length of the optimized filter is 2mM, where
m is the number of lattice section and M is the sub-
bands number.
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
−120
−100
−80
−60
−40
−20
0
20
Frequency (Normalized)
Magnitude(dB)
Figure 3: Magnitude response of the 16-channel IIR filter
bank.
8. Real Time implementation
The equalizer has been implemented as a PlugIn
of the NU-Tech software [7, 13].
NU-Tech is a DSP platform to implement, test,
and tune algorithms in real-time scenarios
through a PC workbench.
It is based on a plug-in architecture which tries to
ease the algorithm writing process.
NU-Tech gives the developer the freedom to write
his own NUTSs (NU-Tech Satellites) in C++, plug
them into the GUI, and test the final result on a
common PC.
A low-level ASIO 2.2 interface allows minimum
and repeatable latencies fully exploiting hardware
resources.
Two PlugIn have been developed in order to bet-
ter exploit the characteristics of the two struc-
tures.
Figure 4: NU-Tech implementation of the FIR equalizer.
Figure 5: NU-Tech implementation of the IIR equalizer.
9. FIR based equalizer
A FIR UFBEq (Uniform Filter Bank Equalizer) NUTs
has been realized as a standard C++ dll file able to
work within the NU-Tech framework.
FIR Filtering procedure
A Partitioned Convolution Overlap and Save algo-
rithm has been implemented [7].
This technique provides an efficient implementa-
tion needed for real time applications, especially
when filters lengths are longer than the working
framesize of the platform.
Fig. 4 shows a uniform filter bank equalizer with
eight bands.
To evaluate the performance, a white noise sig-
nal with flat frequency response has been used:
through the graphical interface it is possible to set
the gain of each individual band.
IIR based equalizer
A IIR UFBEq (Uniform Filter Bank Equalizer) NUTs
has been realized as a standard C++ dll file able to
work within the NU-Tech framework.
IIR Filtering procedure
An optimized function of the IPP libraries [14] has
been considered.
In particular, since the denominator is the same for
each band, this operation is performed separately
in order to optmize the filtering operation.
In Fig. 5, a uniform filter bank equalizer with eight
bands is shown.
As for the FIR equalizer, a white noise signal with
flat frequency response has been used to evaluate
the performance: through the graphical interface it
is possible to set the gain of each individual band.
10. Algorithm validation
Validation
Through a direct comparison between the FIR and the IIR Equalizer implementation [7, 8].
4 cases have been considered with a different number of subbands (i.e.,M = 8, 16, 32, 64), and for
each case:
3 FIR filter prototypes of length N = 1024, 2048, 4096 [Plot (a),(b),(c)];
3 IIR filter prototypes realized with m = 3, 4, 5 sections [Plot (d),(e),(f)].
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
0
1
2
3
4
5
6
7
8
9
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
f
i
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
0
1
2
3
4
5
6
7
8
9
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
f
ii
Figure 6: Equalizer for (i) M = 8, (ii) M = 16.
0.12 0.121 0.122 0.123 0.124 0.125 0.126 0.127 0.128 0.129 0.13
1.8
2
2.2
2.4
2.6
2.8
3
3.2
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
f
i
0.12 0.121 0.122 0.123 0.124 0.125 0.126 0.127 0.128 0.129 0.13
3.8
4
4.2
4.4
4.6
4.8
5
5.2
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
f
ii
Figure 7: Detail of Fig. 8 for (i) M = 8, (ii) M = 16.
11. Algorithm validation
FIR prototype
good performance with a reduced number of
subbands
increasing the number of subbands, the length of
1024 is no more sufficient to have a sharp
transition band: longer FIR prototype are
requested.
IIR prototype
good performance also with a small number of
sections (i.e., m = 3),
it preserve its behaviour increasing the number of
the subbands.
a computational saving with a good performance
level.
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
0
1
2
3
4
5
6
7
8
9
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
f
i
0 0.05 0.1 0.15 0.2 0.25 0.3 0.35 0.4 0.45 0.5
0
1
2
3
4
5
6
7
8
9
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
ii
Figure 8: Equalizer for (i) M = 32 and (ii) M = 64.
0.154 0.1545 0.155 0.1555 0.156 0.1565 0.157 0.1575 0.158 0.1585 0.159
1.8
2
2.2
2.4
2.6
2.8
3
3.2
Frequency (Normalized)
Magnitude(dB)
a
b
c
d
e
f
i
0.155 0.1555 0.156 0.1565 0.157 0.1575
3.8
4
4.2
4.4
4.6
4.8
5
5.2
Frequency (Normalized)
Magnitude(dB) a
b
c
d
e
ii
Figure 9: Detail of Fig. 8 for (i) M = 32 and (ii) M = 64.
12. Algorithm validation
Objective measurements
A distortion index to have a direct comparison be-
tween different equalizer with different number of
bands:
DI =
max T(ejω) + min T(ejω)
2
. (9)
with the distortion transfer function [15], calculated
as follows
T(z) =
1
M
M
i=1
Hi(z)Fi(z) (10)
where Hi and Fi are the frequency responses of the
analysis and synthesis filter banks, respectively.
This index takes into account the amplitude distor-
tion of each band: better performance is achieved
when it is approximately 1.
FIR lengths (N)
M 1024 2048 4096
8 0.997946 0.998508 0.997859
16 0.997946 0.998508 0.997859
32 0.997947 0.998508 0.997859
64 0.997946 0.998508 0.997860
Table 1: Distortion index for uniform M band equalizer with a
FIR prototype of length N.
IIR sections (m)
M 3 4 5
8 0.989940 0.990390 0.990388
16 0.997324 0.995717 0.997588
32 0.999378 0.999398 0.999397
64 0.999773 0.999849 0.999846
Table 2: Distortion index for uniform M band equalizer with a
IIR prototype of length 2nM, where n is the number of lattice
sections.
Good performance for all the considered prototype (i.e., the obtained values are very close to 1).
Better performance (in terms of DI) for the IIR equalizer increasing the subbands number.
The validity of the proposed approach is confirmed in comparison with the first FIR implementation.
13. Conclusions
A new approach to M-band uniform IIR equalizer has been presented
Employment of IIR filters to improve the filter banks structure avoiding ripple between adjacent bands.
Several advantages: increasing the equalization performance in terms of low computational
complexity, low delay, and uniform frequency response.
Objective results as comparison with FIR structure confirm the validity of the proposed approach.
Future works will be oriented to an improvement in the project of the IIR prototype and to the
introduction of polyphase components in the filtering structure, exploiting the lattice structure behaviour.
References
[1] J. Vieira, “Digital Five Band Equalizer with Linear Phase,” in Proc. 100th Audio Engineering Society Convention, Copenhagen, Denmark, May 1996.
[2] Y. Lim, “A Digital Filter Bank for Digital Audio Systems,” IEEE Trans. Circuits Syst., vol. 33, no. 8, pp. 848–849, Aug. 1986.
[3] J. Henriquez, T. Riemer, and R. Trahan Jr., “A Phase Linear Audio Equalizer: Design and Implementation,” J. Audio Eng. Soc., vol. 38, no. 9, pp. 653–666, Sept. 1990.
[4] H. Schopp and H. Hetzel, “A Linear Phase 512 Band Graphic Equalizer using the Fast Fourier Transform,” in Proc. 96th Audio Engineering Society Convention, Amsterdam, The Netherlands, Feb. 1994.
[5] A. Gilloire and M. Vetterli, “Adaptive filtering in subbands with critical sampling: analysis, experiments and application to acoustic echo cancellation,” IEEE Trans. Signal Process., vol. 40, no. 8, pp. 1862–1875,
Aug. 1992.
[6] M. Petraglia and P. Batalheiro, “Non uniform Subband Adaptive Filtering with Critical Sampling,” in Proc. IEEE International Symposium on Circuits and Systems, vol. 56, no. 2, Island of Kos, Greece, May 2006,
pp. 565–575.
[7] S. Cecchi, P. Peretti, L. Palestini, F. Piazza, F. Bettarelli, and A. Lattanzi, “Real Time implementation of an Innovative Digital Audio Equalizer,” in Proc. 123rd Audio Engineering Society Convention, New York, NY,
USA, Oct. 2007.
[8] S. Cecchi, L. Palestini, E. Moretti, and F. Piazza, “A New Approach to Digital Audio Equalization,” in Proc. Workshop on Applications of Signal Processing to Audio and Acoustics, New Paltz, NY, USA, October
2007, pp. 62–65.
[9] R. Koilpillai and P. Vaidyanathan, “Cosine-modulated fir filter banks satisfying perfect reconstruction,” Signal Processing, IEEE Transactions on, vol. 40, no. 4, pp. 770 –783, apr 1992.
[10] S. Kim and C. Yoo, “Highly selective m-channel iir cosine-modulated filter banks,” Electronics Letters, vol. 39, no. 20, pp. 1478 – 1479, oct. 2003.
[11] M. Petraglia, R. Alves, and P. Diniz, “New Structures for Adaptive Filtering in Subbands with Critical Sampling,” IEEE Trans. Signal Process., vol. 48, no. 12, pp. 3316–3327, Dec. 2000.
[12] F. Cruz-Roldan, P. Amo-Lopez, S. Maldonado-Bascon, and S. S. Lawson, “An Efficient and Simple Method for Designing Prototype Filters for Cosine-Modulated Pseudo-QMF Banks,” IEEE Signal Process. Lett.,
vol. 9, no. 1, pp. 29–31, Jan. 2002.
[13] A. Lattanzi, F. Bettarelli, and S. Cecchi, “NU-Tech: The Entry Tool of the hArtes Toolchain for Algorithms Design,” in Proc. 124th Audio Engineering Society Convention, Amsterdam, The Netherlands, May 2008,
pp. 1–8.
[14] “IPP Libraries,” Intel Corporation, 2011. [Online]. Available: http://software.intel.com/en-us/intel-ipp/
[15] P. P. Vaidyanathan, Multirate Systems and Filterbanks. Prentice-Hall, Inc., 1993.