164 research outputs found

    A novel approach to design low-cost two-stage frequency-response masking filters

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    The multistage frequency-response masking (FRM) technique is widely used to reduce the complexity of a filter when the transition bandwidth is extremely small. In this brief, a real generalized two-stage FRM filter without any constraint on the subfilters or the interpolation factors was proposed. New principles and equations were deduced to determine the design parameters. The subfilters were then jointly optimized using non-linear optimization. Experiential results show that when the proposed algorithm obtains different solutions with the conventional algorithm, the solution of the proposed approach is better with less number of filter coefficients and sometimes with lower delay as well than the conventional two-stage FRM, which can lead to a reduced hardware cost in applications

    FRM-Based FIR filters with minimum coefficient sensitivities

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    A method for optimizing FRM-based FIR filters with optimum coefficient sensitivity is presented. This technique can be used in conjunction with nonlinear optimization techniques to design very sharp filters that do not only have very sparse coefficient values but also very low coefficient sensitivity

    Digital Filters and Signal Processing

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    Digital filters, together with signal processing, are being employed in the new technologies and information systems, and are implemented in different areas and applications. Digital filters and signal processing are used with no costs and they can be adapted to different cases with great flexibility and reliability. This book presents advanced developments in digital filters and signal process methods covering different cases studies. They present the main essence of the subject, with the principal approaches to the most recent mathematical models that are being employed worldwide

    Synthesis methods for linear-phase FIR filters with a piecewise-polynomial impulse response

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    his thesis concentrates on synthesis methods for linear-phase finite-impulse response filters with a piecewise-polynomial impulse response. One of the objectives has been to find integer-valued coefficients to efficiently implement filters of the piecewise-polynomial impulse response approach introduced by Saram¨aki and Mitra. In this method, the impulse response is divided into blocks of equal length and each block is created by a polynomial of a given degree. The arithmetic complexity of these filters depends on the polynomial degree and the number of blocks. By using integer-valued coefficients it is possible to make the implementation of the subfilters, which generates the polynomials, multiplication-free. The main focus has been on finding computationally-efficient synthesis methods by using a piecewise-polynomial and a piecewise-polynomial-sinusoidal impulse responses to make it possible to implement high-speed, low-power, highly integrated digital signal processing systems. The earlier method by Chu and Burrus has been studied. The overall impulse response of the approach proposed in this thesis consists of the sum of several polynomial-form responses. The arithmetic complexity depends on the polynomial degree and the number of polynomial-form responses. The piecewise-polynomial-sinusoidal approach is a modification of the piecewise-polynomial approach. The subresponses are multiplied by a sinusoidal function and an arbitrary number of separate center coefficients is added. Thereby, the arithmetic complexity depends also on the number of complex multipliers and separately generated center coefficients. The filters proposed in this thesis are optimized by using linear programming methods

    Design and implementation of computationally efficient digital filters

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    Ph.DDOCTOR OF PHILOSOPH

    Design of Computationally Efficient Digital FIR Filters and Filter Banks

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    Ph.DDOCTOR OF PHILOSOPH

    Digital Filters

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    The new technology advances provide that a great number of system signals can be easily measured with a low cost. The main problem is that usually only a fraction of the signal is useful for different purposes, for example maintenance, DVD-recorders, computers, electric/electronic circuits, econometric, optimization, etc. Digital filters are the most versatile, practical and effective methods for extracting the information necessary from the signal. They can be dynamic, so they can be automatically or manually adjusted to the external and internal conditions. Presented in this book are the most advanced digital filters including different case studies and the most relevant literature

    Source Separation for Hearing Aid Applications

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