Finite impulse response

In signal processing, a finite impulse response (FIR) filter is a filter whose impulse response (or response to any finite length input) is of finite duration, because it settles to zero in finite time. This is in contrast to infinite impulse response (IIR) filters, which may have internal feedback and may continue to respond indefinitely (usually decaying). The impulse response of an Nth-order discrete-time FIR filter (i.e., with a Kronecker delta impulse input) lasts for N + 1 samples, and then settles to zero. FIR filters can be discrete-time or continuous-time, and digital or analog. The output y of a linear time invariant system is determined by convolving its input signal x with its impulse response b. For a discrete-time FIR filter, the output is a weighted sum of the current and a finite number of previous values of the input. The operation is described by the following equation, which defines the output sequence y[n] in terms of its input sequence x[n]:

where: is the input signal, is the output signal, are the filter coefficients, also known as tap weights, that make up the impulse response, is the filter order; an th-order filter has terms on the right-hand side. The in these terms are commonly referred to as taps, based on the structure of a tapped delay line that in many implementations or block diagrams provides the delayed inputs to the multiplication operations. One may speak of a 5th order/6-tap filter, for instance.

Impulse response
The impulse response can be calculated if we set in the above relation, where the Kronecker delta impulse. The impulse response for an FIR filter then becomes the set of coefficients , as follows is

for to . The Z-transform of the impulse response yields the transfer function of the FIR filter

FIR filters are clearly bounded-input bounded-output (BIBO) stable, since the output is a sum of a finite number of finite multiples of the input values, so can be no greater than times the largest value appearing in the input.

Infinite impulse response

Infinite impulse response (IIR) is a property of signal processing systems. Systems with this property are known as IIR systems or, when dealing with filter systems, as IIR filters. IIR systems have an impulse response function that is non-zero over an infinite length of time. This is in contrast to finite impulse response (FIR) filters, which have fixed-duration impulse responses. The simplest analog IIR filter is an RC filter made up of a single resistor (R) feeding into a node shared with a single capacitor (C). This filter has an exponential impulse response characterized by an RC time constant. Because the exponential function is asymptotic to a limit, and thus never settles at a fixed value, the response is considered infinite. IIR filters may be implemented as either analog or digital filters. In digital IIR filters, the output feedback is immediately apparent in the equations defining the output. Note that unlike FIR filters, in designing IIR filters it is necessary to carefully consider the "time zero" case[citation needed] in which the outputs of the filter have not yet been clearly defined. Design of digital IIR filters is heavily dependent on that of their analog counterparts because there are plenty of resources, works and straightforward design methods concerning analog feedback filter design while there are hardly any for digital IIR filters. As a result, usually, when a digital IIR filter is going to be implemented, an analog filter (e.g. Chebyshev filter, Butterworth filter, Elliptic filter) is first designed and then is converted to a digital filter by applying discretization techniques such as Bilinear transform or Impulse invariance. Example IIR filters include the Chebyshev filter, Butterworth filter, and the Bessel filter.

Transfer function derivation

Digital filters are often described and implemented in terms of the difference equation that defines how the output signal is related to the input signal:

where: is the feedforward filter order are the feedforward filter coefficients is the feedback filter order are the feedback filter coefficients is the input signal is the output signal. A more condensed form of the difference equation is:

which, when rearranged, becomes:

To find the transfer function of the filter, we first take the Z-transform of each side of the above equation, where we use the time-shift property to obtain:

We define the transfer function to be:

Considering that in most IIR filter designs coefficient more traditional form:

is 1, the IIR filter transfer function takes the

power spectral density

power spectral density (PSD), which describes how the power of a signal or time series is distributed with frequency. Here, power can be the actual physical power, or more often, for convenience with abstract signals, can be defined as the squared value of the signal. This instantaneous power is then given by for a signal . The mean (or expected value) of is the total power, which is the integral of the power spectral density over all frequencies. We can use a normalized Fourier transform:

and define the power spectral density as:[4][5] Fourier transform in a formal way to formulate a definition of the power spectral density , where is the Dirac delta function. Such formal statements may be sometimes useful to guide the intuition, but should always be used with utmost care. The power spectral density and the autocorrelation function of this signal , should be a Fourier pair.

The power of the signal in a given frequency band [w1,w2] can be calculated by integrating over positive and negative frequencies,

The power spectral density of a signal exists if the signal is a wide-sense stationary process. If the signal is not wide-sense stationary, or strictly stationary, then the autocorrelation function must be a function of two variables. In some cases, such as wide-sense cyclostationary processes, a PSD may still exist.More generally, similar techniques may be used to estimate a time-varying spectral density. The definition of the power spectral density generalizes in a straightforward manner to finite time-series with for a total measurement period such as a signal sampled at discrete times .

. If two signals both possess power spectra (the correct terminology), then a cross-power spectrum can be calculated by using their cross-correlation function. "FFT" redirects here. For other uses, see FFT (disambiguation).

Fast Fourier transform

A fast Fourier transform (FFT) is an efficient algorithm to compute the discrete Fourier transform (DFT) and its inverse. There are many distinct FFT algorithms involving a wide range of mathematics, from simple complex-number arithmetic to group theory and number theory; this article gives an overview of the available techniques and some of their general properties, while the specific algorithms are described in subsidiary articles linked below. A DFT decomposes a sequence of values into components of different frequencies. This operation is useful in many fields (see discrete Fourier transform for properties and applications of the transform) but computing it directly from the definition is often too slow to be practical. An FFT is a way to compute the same result more quickly: computing a DFT of N points in the naive way, using the definition, takes O(N2) arithmetical operations, while an FFT can compute the same result in only O(N log N) operations. The difference in speed can be substantial, especially for long data sets where N may be in the thousands or millionsin practice, the computation time can be reduced by several orders of magnitude in such cases, and the improvement is roughly proportional to N / log(N). This huge improvement made many DFT-based algorithms practical; FFTs are of great importance to a wide variety of applications, from digital signal processing and solving partial differential equations to algorithms for quick multiplication of large integers. The best-known FFT algorithms depend upon the factorization of N, but there are FFTs with O(N log N) complexity for all N, even for prime N. Many FFT algorithms only depend on the fact that is an th primitive root of unity, and thus can be applied to analogous transforms over any finite field, such as number-theoretic transforms. Since the inverse DFT is the same as the DFT, but with the opposite sign in the exponent and a 1/N factor, any FFT algorithm can easily be adapted for it. The FFT has been described as "the most important numerical algorithm of our lifetime".[1 An FFT computes the DFT and produces exactly the same result as evaluating the DFT definition directly; the only difference is that an FFT is much faster. (In the presence of round-off error, many FFT algorithms are also much more accurate than evaluating the DFT definition directly, as discussed below.) Let x0, ...., xN-1 be complex numbers. The DFT is defined by the formula