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                         If a signal has periodic content, but is not periodic, such as x[n] = cos(0.5πn) + cos(0.2n), then leakage
                       cannot be avoided by a selection of N. An alternate means of reducing leakage is to first taper the signal
                       to zero at the initial and end points of the sequence prior to computing the DFT. This process, known
                       as  windowing the data, is accomplished by multiplying  x[n] by a window function  w[n] and then
                       performing the DFT on the product x[n]w[n]. Three common windows are the rectangular window,
                       which is a sharp truncation, the Hanning window, and the Hamming window [1].


                         Rectangular Window:
                                                   w[n] = 1,  0 ≤ n ≤ N −  1

                         Hanning Window:

                                                                 
                                                             2pn
                                            wn[] =  1   cos  ------------- ,0 ≤ n ≤  N 1
                                                    -- 1 –
                                                                             –
                                                    2      N 1
                                                              –
                         Hamming Window:
                                                                   
                                                               2pn
                                           wn[] =  0.54 0.46cos   ------------- , 0 ≤ n ≤  N 1
                                                      –
                                                                               –
                                                               N 1
                                                                 –
                         If the value of N in a DFT is a power of 2, there is a fast method to compute the DFT called the fast
                       Fourier transform (FFT). If the value of N is not a power of 2, zeros can be padded to the end of the
                       signal in order to use the FFT. This does not affect the accuracy of the result, but it does improve the
                       resolution of the resulting plot when the DFT (or FFT) is used to compute the DTFT. In many cases,
                       the expression used in Eq. (29.4) suffices to compute the DFT since the added computational power of
                       today’s processors lessens the need for the numerical efficiency of the FFT. The details of the algorithm
                       for the FFT are beyond the scope of this handbook. See [1] or [2] for details.

                       29.3 Continuous-Time to Discrete-Time Mappings

                       While most physical systems operate in continuous-time, computers operate in discrete-time. Therefore,
                       in order to use computers to process measurements taken from continuous-time systems, there must be
                       ways of mapping between the continuous-time world to the discrete-time world.

                       Discretization
                       Before an analog signal can be analyzed using digital techniques, it must be discretized (that is, converted
                       into a discrete-time signal). The ideal method for discretization is sampling, where the values of the signal
                       are determined at discrete points in time. Generally, the signal is sampled at a fixed rate known as the sampling
                       period. The sampling rate (in hertz) is the inverse of the sampling period. Figure 29.2 depicts a 1 Hz signal
                       that has been sampled at two rates. The dark points are sampled at 15 ms intervals, while the lighter points
                       are sampled at 250 ms intervals. From Fig. 29.2, the waveform approximation clearly degrades as the sampling
                       frequency is reduced and approaches the signal frequency. In fact, it can be shown that a signal must be
                       sampled at a frequency that is higher than twice its maximum frequency content. This is known as the Nyquist
                       Sampling Theorem. For example, if the signal in Fig. 29.2 is sampled at 0.5 Hz, it is possible for every sample
                       to have a value of 0 as at 0, 500, 1000 ms, etc… the value of the signal is 0. The erroneous interpretation of
                       the signal due to a sample frequency that is too low is known as aliasing. There are two means by which the
                       Nyquist Sampling Theorem can be satisfied. The first is by employing a sample frequency that is more than
                       twice the highest frequency content of the signal being sampled. This value frequency is known as the Nyquist
                       frequency. As one never is sure of the actual frequency content of a real signal, a low pass filter may be used
                       to ensure that a signal does not possess frequencies above a certain cut-off level. Such a filter is commonly


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