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a digital filter is a system that performs mathematical operations on a sampled, discrete-time signal to reduce or enhance certain aspects of that signal A digital filter system usually consists of an analog-to-digital converter to sample the input signal, followed by a microprocessor and some peripheral components such as memory to store data and filter coefficients etc. Finally a digital-to-analog converter to complete the output stage.

sinusoidal response

Digital filters are often best described in terms of their frequency response. That is, how is a sinusoidal signal of a given frequency affected by the filter. Below is the very short summary of sinusoidal response of digital filters detailed explanation could be seen in text book or internet.

Before a signal is applied to the input of a digital filter, the filter's internal ``state'' is assumed equal to zero. Digital filters are linear systems. One property of linear systems is that a sinusoidal input will produce a sinusoidal output of the same frequency. However, when a sinusoidal signal is first applied to the input of a digital filter, the output initially exhibits a region of transition (referred to as its transient response). For FIR filters, this transition region (or ``warm-up'' period) has a duration in samples equal to the filter order. For IIR filters, the length of the transition region is dependent on the filter order and the feedback coefficient values. Assuming a continued application of the sinusoidal input, the filter will eventually settle into its steady-state region. If the input changes frequency or displays a discontinuity of any sort, another transient region will occur in the filter output. The frequency response of a digital filter is understood to represent its steady-state behavior.

1. IIR is infinite and used for applications where linear characteristics are not of concern. 2. FIR filters are Finite IR filters which are required for linear-phase characteristics. 3. IIR is better for lower-order tapping, whereas the FIR filter is used for higher-order tapping.

4. FIR filters are preferred over IIR because they are more stable, and feedback is not involved. 5. IIR filters are recursive and used as an alternate, whereas FIR filters have become too long and cause problems in various applications.

Below is the summary of of basic iir filter pole zero design.

The z-transform is extensively used to evaluate the properties of discrete-time systems such as digital filters. In particular, it is convenient for determining the stability of a system. The numerator and denominator of a system's transfer function are polynomials in z. The roots of these polynomials can be determined by factorization. Roots of the numerator polynomial indicate values of z at which the transfer function evaluates to zero. These are called zeros. Roots of the denominator polynomial indicate values of z at which the transfer function evaluates to infinity. These are called poles. The zeros and poles of a transfer function can be plotted in the z-plane. Their locations with respect to the unit circle indicate radian frequencies at which the system's magnitude response has local minima (near zeros) or maxima (near poles). When the coefficients of a transfer function are all real, complex roots are given by complex-conjugate pairs. For a system to have a stable frequency response, all of its poles must lie within the unit circle in the z-plane.

In signal processing, a window function (also known as an apodization function or tapering function[1]) is a mathematical function that is zero-valued outside of some chosen interval.

The design of a FIR filter starts with its specifications in either discrete-time domain or DTFT frequency domain, or both. In the time domain, the design objective is the impulse response. In the frequency domain, the requirement is on various parameters of the magnitude response Overall filter design From the filter specifications, the first step is to choose between FIR and IIR filters based on their advantages and disadvantages. This chapter only concerns FIR filters. The next step is to

select the proper linear phase FIR filters (section 5.2.3). A rather complete procedure for the design of FIR filters are as follows. * Specifications of the filter * Choosing an appropriate linear phase filter type * Choosing the mothod of design such as window, optimal, frequency sampling * Calculation the filter coefficients (impulse response) * Finding suitable structure * Analysis of the finite wordlength effects * Implementation of the filter in hardware and/or software

Retangular window

The rectangular window is sometimes known as a Dirichlet window. It is the simplest window, equivalent to replacing all but N values of a data sequence by zeros, making it appear as though the waveform suddenly turns on and off. Other windows are designed to moderate the sudden changes because discontinuities have undesirable effects on the discrete-time Fourier transform (DTFT) and/or the algorithms that produce samples of the DTFT Hamming window

The "raised cosine" with these particular coefficients was proposed by Richard W. Hamming. The window is optimized to minimize the maximum (nearest) side lobe, giving it a height of about one-fifth that of the Hann window, a raised cosine with simpler coefficients.[10][11]

[note 1]

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