DSP Notes: Sampling and Transforms

Professor Fred DePiero, CalPoly State University

Important Points on Sampling
1) The Sampling Theorem states that a signal can be sampled without loss of
information, provided the sampling rate, Fs (Hz), is at or above the Nyquist rate
Fs >= 2B
where B is the bandwidth of the signal.
2) Sampling a signal x(t) changes its spectrum, X(F). The new spectrum contains
multiple copies of the original version, X(F). See page 272 in the Proakis text. The
copies of the original spectrum appear centered at multiples of Fs. This replication is
due to the fact that sampling is equivalent to multiplication by an impulse train, d(t),
having impulses separated by 1/Fs (sec). The spectrum of d(t) is another impulse train,
D(F), with impulses separated by Fs (Hz). Hence the action of multiplying by d(t) in the
time domain is equivalent to convolution with D(F). The convolution operation results in
the spectral copies.
3) When a signal with an analog frequency F0 is digitized at a sample rate Fs < 2 F0,
then aliasing occurs. In this case, the frequency of the digitized version of the signal is
lower than the original analog version. For a signal with bandwidth B, as in (1) and (2)
above, aliasing occurs when Fs < 2B and results in overlapping copies of X(F). Some
systems use aliasing in a beneficial way. For example, a baseband signal can be
recovered from a modulated signal through the process of downconversion, which can
be implemented using aliasing.
4) The relationship between the normalized frequency of a digitized signal, w0 (radians
/ sample), and the original analog frequency, F0 (Hz), is
w0 = 2 pi F0 / Fs
This relation is true, irrespective of any aliasing. An example of a discrete signal at this
frequency is x(n) = cos(w0 n). The samples of x(n) occur at times t = n dT = n / Fs, dT is
the sample spacing (sec).
5) For a given sampling frequency, Fs, the highest frequency analog signal that can be
digitized without aliasing is at
F0 = Fs / 2 = Fd (Hz)

<=>

w0 = pi (radians / sample)

Analog signals at higher frequencies cant be appreciated as such in the discrete

domain. When dealing with discrete signals, the most useful range of the frequency axis
is
-Fs / 2 <= f <= Fs / 2 or -pi <= w <= pi
The spectrum outside this range contains redundant information, due to sampling.
Important Points on Various Transforms of Discrete Signals

1) Discrete-Time Fourier Transform (DTFT) yields the spectrum X(w) of a discrete time
signal, x(n). X(w) is periodic with period = 2 pi and the units of w are radians / sample.
Typically the range of: -pi <= w <= pi is most useful because a sinusoid at w0 = pi
cos( w0 n ) = cos( pi n ) = (-1)^n
is at the highest frequency possible for a discrete signal. (See discussion on sampling.)
X(w) is generally evaluated by hand, in that it involves symbolic manipulation of
functions. Hence it is not considered to be computationally feasible. It is a continuous
function. The DTFT is appropriate for use in hand calculations when the signal x(n) is
known for all time and is expressed in analytic form, such as forms involving sin(),
cos(), exp(), or polynomials, for example. This is not possible for real-world signals that
are digitized and known only in terms of a finite set of samples.
2) The Discrete Fourier Transform, DFT, is a computationally feasible version of the
Fourier Transform. It is designated:
x(n) <= DFT, N => X(k)
where x(n) are samples in the time domain and X(k) are samples in the frequency
domain. There are N samples of both x(n) and X(k). X(k) are samples of the continuous
function. X(w). These samples X(k) occur at frequencies
w = k dw (radians / sample)
where dw is the spacing between samples in the frequency domain.
dw = 2 pi / N (radians / sample)

<=>

dF = Fs / N = 1 / N dT = 1 / L (Hz)

where L is the length of the signal (sec). Since X(k) contains only samples in the
frequency domain, some details of a signals true spectrum, X(w), may not be apparent
in X(k).
3) The Fast Fourier Transform, FFT, refers to a family of algorithms which yield the
exact same result as the DFT, but are computationally efficient. The most common kind
of FFT is the radix-2 type. For this type N = 2^r.
4) The spectrum of a version of a signal having finite duration differs from the spectrum
of its infinite duration version. This is referred to as the spectral leakage effect. For
example, given the signal
x(n) = cos(w0 n)
with infinite duration. Its spectrum X(w), -pi/2 <= w <= pi/2, consists of a pair of
impulses at +/- w0. The finite duration version of this signal can be viewed as a
multiplication of x(n) by a rectangular window function, w(n). This is equivalent to
convolution with a sinc function in the frequency domain. This convolution spreads out
the energy contained in the original two impulses resulting in the spectral leakage
phenomena.
This effect is reduced by having a window w(n) with long duration. For a window of
length L (sec), the zeros of the sinc are separated by 1 / L (Hz). Hence a longer window
results in a narrower sinc and less spectral corruption.

5) The spectral leakage associated with a finite version of an infinite duration signal will
always be apparent in X(w), as found via the DTFT. Because the DFT operates on
versions of signals having finite duration, it naturally causes spectral leakage to occur.
However, spectral leakage may not be evident due to the discrete sampling introduced
by X(k). It will not be evident if, for example, (a) the signal is a single harmonic at F0
where F0 is an integer multiple of dF, and (b), if the spacing between the zeros of the
sinc function (associated with spectral leakage) exactly equals the spectral spacing. If
there is no zero padding of the signal then conditions (a) and (b) amount to the
requirement that
F0 N / Fs = c
where c is any integer. So to determine if spectral leakage will not be observed, compute
c, as above. If c is an integer then leakage will not be observed - regardless of the
duration of the signal, N.
6) Zero padding is the process of concatenating zeros to the end of a signal. This
operation may be performed on any signal, x(n), prior to computing its transform with a
DFT. Zero padding will cause X(k) to approach X(w). Zero padding helps to reveal
spectral leakage effects.
7) Zero padding has additional uses. In order to derive the DFT it was necessary to
make the assumption that x(n) is periodic outside the interval of [0, N-1]. This has an
important implication when evaluating a time-domain convolution x(n)*y(n) by a
frequency-domain multiplication of the spectrums X(k) Y(k), as computed via the DFT.
The point of concern arises because the frequency-domain multiplication corresponds
to convolution of periodic versions of x(n) and y(n). This operation is referred to as
periodic convolution. In general x(n) and y(n) are not periodic in N. In these cases a
technique is needed to yield the same result as linear convolution would, when
evaluated in the time-domain. If x(n) has length L and y(n) has length M, then a linear
convolution can be achieved by first zero padding x(n) and y(n) to a length N:
N = L + M - 1.