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The Butterworth Filter is the maximally flat amplitude filter. It provides a near 0 attenuation until near the cutoff frequency and then descends into attenuation smoothly. The tranition becomes sharper with higher orders. It has moderate group delay so it has some overshoot on sharp rising waveforms. this gets worse with higher orders. The Chebyshev filter trades off flatness in the pass band for a steeper decline into the stop band. You design a cheby with a recurring wavelike ripple of attenuation in the paasband of usually 0.05db to 3db. In return you get a much steeper portion of the attenuation curve near the cutoff frequency. waveforms are distorted by group delay errors more severely than in the butterworth. The higher the ripple the worse the distortion The elliptical filter is like cheby^2 since it has ripples in the stopband and the passband. It has been proven to be the fastest possible descent into the stopband. The time delay errors are more severe than the cheby. For all of these filters. when at a distance from the cutoff frequency they will attenuate 20db/decade in frequency times the filter order.

1) The unit impulse is a signal which is zero everywhere except when its argument is zero, then it is equal to 1. Mathematically

The unit sample function is very useful in that it can be seen as the elementary constituent in any discrete signal. Let

x(n)

be a sequence. Then

we can express

delay operation)

2) The unit step function is equal to zero when its index is negative and equal to one for non-negative indexes, see Figure 1 for plots.

UNIT STEP:

The ramp function, denoted by r ( t ) is a signal whose amplitude increases proportionally as time increases. The mathematical definition of ramp signal is Fig.shows a ramp signal of the function A discrete-time form a ramp signal is called a ramp sequence and is shown in Fig.

Analysis of differential equations Fourier transforms and the closely related Laplace transforms are widely used in solving differential equations. The Fourier transform is compatible with differentiation in the following sense: if f(x) is a differentiable function with Fourier transform , then the Fourier transform of its derivative is given by. This can be used to transform differential equations into algebraic equations. Note that this technique only applies to problems whose domain is the whole set of real numbers. By extending the Fourier transform to functions of several variables partial differential equations with domain Rn can also be translated into algebraic equations. Fourier transform spectroscopy The Fourier transform is also used in nuclear magnetic resonance (NMR) and in other kinds of spectroscopy, e.g. infrared (FTIR). In NMR an exponentially-shaped free induction decay (FID) signal is acquired in the time domain and Fourier-transformed to a Lorentzian line-shape in the frequency domain. The Fourier transform is also used in magnetic resonance imaging (MRI) and mass spectrometry.

A Discrete Fourier Transform is simply the name given to the Fourier Transform when it is applied to digital (discrete) rather than an analog (continuous) signal. An FFT (Fast Fourier Transform) is a faster version of the DFT that can be applied when the number of samples in the signal is a power of two. An FFT computation takes approximately N * log2(N) operations, whereas a DFT takes approximately N^2 operations, so the FFT is significantly faster

The Fourier series is important because it allows one to model periodic signals as a sum of distinct harmonic components. In other words, representing signals in this way allows one to see the harmonics in a signal distinctly, which makes it easy to see what frequencies the signal contains in order to filter/manipulate particular frequency components.

In mathematics, the continuous Fourier transform is one of the specific forms of Fourier analysis. As such, it transforms one function into another, which is called the frequency domain representation of the original function (which is often a function in the time-domain). In this specific case, both domains are continuous and unbounded. The term Fourier transform can refer to either the frequency domain representation of a function or to the process/formula that "transforms" one function into the other.

Below I make assumptions and generalizations about message board and weblog design. My goal is to discuss what I think are standard practices across the technologies. I realize that the

assumptions below may or may not match with your experiences and I present them as suggestions. Please comment or email me with any input. First, I believe that weblogs and message boards *are* different -different enough to happily exist together in the same online community web site. My conclusion is that online communities will use the two resources to fill two different roles. Their ability to fill independent niches will make the subtle differences between them make more sense. The table below outlines the differences I see. Below the table is a description of each row.

Q7 difference between linear and circular convolution????? Linear convolution takes two functions of an independent variable, which I will call time, and convolves them using the convolution sum formula you might find in a linear sytems or digital signal processing book. Basically it is a correlation of one function with the time-reversed version of the other function. I think of it as flip, multiply, and sum while shifting one function with respect to the other. This holds in continuous time, where the convolution sum is an integral, or in discrete time using vectors, where the sum is truly a sum. It also holds for functions defined from -Inf to Inf or for functions with a finite length in time. Circular convolution is only defined for finite length functions (usually, maybe always, equal in length), continuous or discrete in time. In circular convolution, it is as if the finite length functions repeat in time, periodically. Because the input functions are now periodic, the convolved output is also periodic and so the convolved output is fully specified by one of its periods. Explicit circular convolution is rarely used; however implicitly it is used very frequently. Any time DFTs (FFT) or Fourier Series are multiplied, there is an underlying circular convolution taking place. To make this more concrete, let me leave you with a simple example (in pseudo-MATLAB notation). Here I will use the FFT to compute the circular convolution. x = [1 2 2]; y = [1 3 1]; conv(x,y) == [1 5 9 8 2]; % Notice the result has tails where x and y do not fully overlap X = fft(x);

Y = fft(y); ifft(X*Y) == [9 7 9]; % No tails and the result is different because x and y are assumed periodic % Now if we zero pad x and y to avoid the overlap between one period and the next, then the % circular convolution is the same as the original linear convolution. Though strictly speaking % the circular convolution is infinite-length and periodic, whereas the linear convolution is finite% length. x0 = [1 2 2 0 0]; y0 = [1 3 1 0 0]; X = fft(x); Y = fft(y); ifft(X*Y) == [1 5 9 8 2];

Correlation determines the degree of similarity between two signals. If the signals are identical, then the correlation coefficient is 1; if they are totally different, the correlation coefficient is 0, and if they are identical except that the phase is shifted by exactly 180degree (i.e. mirrored), then the correlation coefficient is -1. When two independent signals are compared, the procedure is known as crosscorrelation, and when the same signal is compared to phase shifted copies of itself, the procedure is known as autocorrelation. A function which is related to the correlation function, but arithmetically less complex, is the average magnitude difference function. Cross-correlation is the method which basically underlies implementations of the Fourier transformation: signals of varying frequency and phase are correlated with the input signal, and the degree of correlation in terms of frequency and phase represents the frequency and phase spectrums of the input signal. Autocorrelation is a method which is frequently used for the extraction of fundamental frequency, :if a copy of the signal is shifted in phase, the distance between correlation peaks is taken to be the fundamental period of the signal (directly related to the fundamental frequency). The method may be combined with the simple smoothing operations of peak and centre clipping, or with other low-pass filter operations.

Sampling can be done for signals varying in space, time, or any other dimension, and similar results are obtained in two or more dimensions. For signals that vary with time, let x(t) be a continuous signal to be sampled, and let sampling be performed by measuring the value of the continuous signal every T seconds, which is called the sampling interval. Thus, the sampled signal x[n] given by:

x[n] = x(nT), with n = 0, 1, 2, 3, ... The sampling frequency or sampling rate fs is defined as the number of samples obtained in one second, or fs = 1/T. The sampling rate is measured in hertz or in samples per second. It is possible under some circumstances to reconstruct the original signal completely and exactly (perfect reconstruction). The NyquistShannon sampling theorem provides a sufficient (but not always necessary) condition under which perfect reconstruction is possible. The sampling theorem guarantees that bandlimited signals (i.e., signals which have a maximum frequency) can be reconstructed perfectly from their sampled version, if the sampling rate is more than twice the maximum frequency. Reconstruction in this case can be achieved using the WhittakerShannon interpolation formula. The frequency equal to one-half of the sampling rate is therefore a bound on the highest frequency that can be unambiguously represented by the sampled signal. This frequency (half the sampling rate) is called the Nyquist frequency of the sampling system. Frequencies above the Nyquist frequency fN can be observed in the sampled signal, but their frequency is ambiguous. That is, a frequency component with frequency f cannot be distinguished from other components with frequencies NfN + f and NfN f for nonzero integers N. This ambiguity is called aliasing. To handle this problem as gracefully as possible, most analog signals are filtered with an anti-aliasing filter (usually a low-pass filter with cutoff near the Nyquist frequency) before conversion to the sampled discrete representation.

Filter Order The maximum delay, in samples, used in creating each output sample is called the order of the filter. In the difference-equation representation, the order is the larger of a in Eq.. For example, specifies a particular second-order filter. If and in Eq. are constrained to be finite (which is, of course, necessary in practice), then Eq.represents the class of all finiteorder causal LTI digital filters.

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