749 research outputs found

    Symbol Synchronization for SDR Using a Polyphase Filterbank Based on an FPGA

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    This paper is devoted to the proposal of a highly efficient symbol synchronization subsystem for Software Defined Radio. The proposed feedback phase-locked loop timing synchronizer is suitable for parallel implementation on an FPGA. The polyphase FIR filter simultaneously performs matched-filtering and arbitrary interpolation between acquired samples. Determination of the proper sampling instant is achieved by selecting a suitable polyphase filterbank using a derived index. This index is determined based on the output either the Zero-Crossing or Gardner Timing Error Detector. The paper will extensively focus on simulation of the proposed synchronization system. On the basis of this simulation, a complete, fully pipelined VHDL description model is created. This model is composed of a fully parallel polyphase filterbank based on distributed arithmetic, timing error detector and interpolation control block. Finally, RTL synthesis on an Altera Cyclone IV FPGA is presented and resource utilization in comparison with a conventional model is analyzed

    Modified DA based FIR Filter in Multirate DSP systems on FPGA

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    Multirate systems are popular in DSP.Systems which employ multiple sampling rates in the processing of digital signals are called Multirate DSP systems, which are used in audio, video processing and communication systems. Multirate DSP systems that employ different chips for different frequency signal results in more area and power utilization. The setback can be avoided by implementing Multirate system, based on Distributed Arithmetic FIR filter. Using such systems, we can achieve computation efficiency and improve the system performance. Modified DA based FIR Filter using Multirate systems includes decimation, interpolation process implemented on FPGA with 53% less LUT utilization compared to existing Multirate system

    Algorithms and architectures for the multirate additive synthesis of musical tones

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    In classical Additive Synthesis (AS), the output signal is the sum of a large number of independently controllable sinusoidal partials. The advantages of AS for music synthesis are well known as is the high computational cost. This thesis is concerned with the computational optimisation of AS by multirate DSP techniques. In note-based music synthesis, the expected bounds of the frequency trajectory of each partial in a finite lifecycle tone determine critical time-invariant partial-specific sample rates which are lower than the conventional rate (in excess of 40kHz) resulting in computational savings. Scheduling and interpolation (to suppress quantisation noise) for many sample rates is required, leading to the concept of Multirate Additive Synthesis (MAS) where these overheads are minimised by synthesis filterbanks which quantise the set of available sample rates. Alternative AS optimisations are also appraised. It is shown that a hierarchical interpretation of the QMF filterbank preserves AS generality and permits efficient context-specific adaptation of computation to required note dynamics. Practical QMF implementation and the modifications necessary for MAS are discussed. QMF transition widths can be logically excluded from the MAS paradigm, at a cost. Therefore a novel filterbank is evaluated where transition widths are physically excluded. Benchmarking of a hypothetical orchestral synthesis application provides a tentative quantitative analysis of the performance improvement of MAS over AS. The mapping of MAS into VLSI is opened by a review of sine computation techniques. Then the functional specification and high-level design of a conceptual MAS Coprocessor (MASC) is developed which functions with high autonomy in a loosely-coupled master- slave configuration with a Host CPU which executes filterbanks in software. Standard hardware optimisation techniques are used, such as pipelining, based upon the principle of an application-specific memory hierarchy which maximises MASC throughput

    Power Estimation Technique for DSP Architectures.

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    The main goal of power estimation is to optimize the power consumption of a electronic design. Power is a strongly pattern dependent function. Input statistics greatly influence on average power. We solve the pattern dependence problem for intellectual property (IP) designs. In this paper, we present a power macro-modeling technique for digital signal processing (DSP) architectures in terms of the statistical knowledge of their primary inputs. During the power estimation procedure, the sequence of an input stream is generated by a genetic algorithm using input metrics. Then, a Monte Carlo zero delay simulation is performed and a power dissipation macro-model function is built from power dissipation results. From then on, this macro-model function can be used to estimate power dissipation of the system just by using the statistics of the macro-block’s primary in puts. In experiments with the DSP system, the average error is 26%

    Design and Implementation of Low Complexity Reconfigurable Filtered-OFDM based LDACS

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    L-band Digital Aeronautical Communication System (LDACS) aims to exploit vacant spectrum in L-band via spectrum sharing, and orthogonal frequency division multiplexing (OFDM) is the currently accepted LDACS waveform. Recently, various works dealing with improving the spectrum utilization of LDACS via filtering/windowing are being explored. In this direction, we propose an improved and low complexity reconfigurable filtered OFDM (LRef-OFDM) based LDACS using novel interpolation and masking based multi-stage digital filter. The proposed filter is designed to meet the stringent non-uniform spectral attenuation requirements of LDACS standard. It offers significantly lower complexity as well as higher transmission bandwidth than state-of-the-art approaches. We also integrate the proposed filter in our end-to-end LDACS testbed realized using Zynq System on Chip and analyze the performance in the presence of LL-band legacy user interference as well as LDACS specific wireless channels. Via extensive experimental results, we demonstrate the superiority of the proposed LRef-OFDM over OFDM and Filtered-OFDM based LDACS in terms of power spectral density, bit error rate, implementation complexity, and group delay parameters.Comment: Paper with Appendi

    FPGA based Uniform Channelizer Implementation

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    Channelizers are widely used in modern digital communication systems. Advanced uniform multirate channelization have been theoretically proved to be capable of reducing the computational load, with a better performance. Therefore, in this thesis, we implement these designs on a FPGA board for the sake of the comprehensive evaluation of resource usage, performance and frequency response. The uniform filter-banks are one of the most essential unit in channelization. The Generalised Discrete Fourier Transform Modulated Filter Bank (GDFT-FB), as an important variant of basic a DFT-FB, has been implemented in FPGA and demonstrated with a better computational saving rather than traditional schemes. Moreover the oversampling version is demonstrated to have a better frequency response with an acceptable amount of extra resources. On the other hand, frequency response masking (FRM) techniques is able to reduce the number of coefficients. Therefore, the full FRM GDFT-FB and alternative narrowband FRM GDFT-FB are both implemented in FPGA platform, in order to achieve a better performance and hardware efficiency

    Low energy HEVC and VVC video compression hardware

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    Video compression standards compress a digital video by reducing and removing redundancy in the digital video using computationally complex algorithms. As spatial and temporal resolutions of videos increase, compression efficiencies of video compression algorithms are also increasing. However, increased compression efficiency comes with increased computational complexity. Therefore, it is necessary to reduce computational complexities of video compression algorithms without reducing their visual quality in order to reduce area and energy consumption of their hardware implementations. In this thesis, we propose a novel technique for reducing amount of computations performed by HEVC intra prediction algorithm. We designed low energy, reconfigurable HEVC intra prediction hardware using the proposed technique. We also designed a low energy FPGA implementation of HEVC intra prediction algorithm using the proposed technique and DSP blocks. We propose a reconfigurable VVC intra prediction hardware architecture. We also propose an efficient VVC intra prediction hardware architecture using DSP blocks. We designed low energy VVC fractional interpolation hardware. We propose a novel approximate absolute difference technique. We designed low energy approximate absolute difference hardware using the proposed technique. We propose a novel approximate constant multiplication technique. We designed approximate constant multiplication hardware using the proposed technique. We quantified computation reductions achieved by the proposed techniques and video quality loss caused by the proposed approximation techniques. The proposed approximate absolute difference technique and approximate constant multiplication technique cause very small PSNR loss. The other proposed techniques cause no PSNR loss. We implemented the proposed hardware architectures in Verilog HDL. We mapped the Verilog RTL codes to Xilinx Virtex 6 or Xilinx Virtex 7 FPGAs and estimated their power consumptions using Xilinx XPower Analyzer tool. The proposed techniques significantly reduced power and energy consumptions of these FPGA implementation

    FPGA implementation of digital timing recovery in software radio receiver

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    This paper describes an implementation of an all-digital timing recovery scheme. Squaring nonlinearity is employed to generate the timing estimate and an IIR is used to extract the spectral component at symbol rate. Hardware design is performed using VHDL and realized in FPGA. The whole design can be fitted into an Altera EPF1OK70 FPGA chip, with 95.5% utilization of logic elements and 22% utilization of memory bits. The implementation exploits features of FPGA, which enable easy implementation of look up table and variable data precision at different nodes.published_or_final_versio
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