Chapter 3 Analysis and Transmission of Signals Students

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EE 23353 Analog Communications

Chapter 3: Analysis and Transmission of Signals
Dr. Rami A. Wahsheh

Communications Engineering Department
Chapter 3: Analysis and Transmission of Signals

3.1 Aperiodic signal representation by Fourier integral. 3.2 Transforms of some useful functions. 3.3 Some properties of the Fourier transform. 3.4 Signal transmission through a linear system. 3.5 Ideal and practical filters. 3.6 Signal distortion over a communication channel. 3.7 Signal energy and energy spectral density. 3.8 Signal power and power spectral density.
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Fourier Transform
The motivation for the Fourier transform comes from the study of Fourier series. In Fourier series complicated periodic functions are written as the sum of simple waves mathematically represented by sines and cosines. Due to the properties of sine and cosine it is possible to recover the amount of each wave in the th sum b an i t by integral. l In many cases it is desirable to use Euler's formula, which states that ei2=cos2+isin2, to write Fourier series in terms of the basic 3 waves ei2.
Fourier Transform
From sines and cosines to complex exponentials makes it necessary for the Fourier coefficients to be complex valued. Complex number gives both the amplitude (or size) of the wave present in the function and the phase (or the initial angle) of the wave. The Fourier series can only b used f Th F i i l be d for periodic i di signals.
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Fourier Transform
How can the results be extended for Aperiodic signals such as g(t) of limited length T ?
3.1 Aperiodic Signal Representation by Fourier Integral

First: Construct a new periodic signal gTo(t) formed by repeating the signal g(t) every To seconds seconds. To is made long enough to avoid overlapping between the repeating pulses
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Second: Calculate the exponential Fourier series of gT (t) with o=2/To 2/T To
Third: Let the period

Integrating gTo(t) over (-To/2,To/2) is the same as integrating g(t) over (-,) therefore ( ),
Observe that the nature of the spectrum changes as To increases. Let us define G(w); a g ( ) continuous function of Then
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Fourier coefficients Dn are (1/To times) the samples of G() uniformly spaced at o rad/sec. Therefore (1/To) G() is the envelope for the coefficients Dn.

Let To by doubling To repeatedly. Doubling To halves the fundamental frequency o, so that there are now twice as many samples in the spectrum.
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If we continue doubling To repeatedly, the spectrum becomes denser while its magnitude becomes smaller, but the relative shape of the envelope will remain the same.
Dn 0
To
wo 0
Spectral components are spaced p p p at zero (infinitesimal) interval
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Fourier Transform
Then Fourier series can be expressed as:
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Fourier Transform
gTo(t) can be expressed as a sum of everlasting exponentials of frequencies
The sum on the right-hand side can be viewed as the area under the function G(w)ejt
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Fourier Transform
The sum on the right-hand side can be viewed as the area under the function G(w)ejt
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Fourier Transform
G(w) is the direct Fourier transform of g(t) g(t) is the inverse F ( ) h Fourier transform of G( ) f f G(w)
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Fourier Transform
G(w) is complex. To plot the spectrum G(w) as a function of , we have both amplitude and phase spectra:
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Conjugate Symmetry Property

If g(t) is a real function of t Then G( ) and G( ) are complex conjugates: Th G() d G(-) l
For real g(t), The amplitude spectrum is an even function The phase spectrum g()17is an odd function of .
Example 3.1
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Example 3.1
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Example 3.1
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Linearity of the Fourier Transform
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3.2 Transforms of Some Useful Functions

A unit gate function rect(x) has a unit height and unit width centered at the origin.
Expanded by
The denominator of the argument of rect(x/) is the 22 width of the pulse.
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Unit Triangle Function

A unit triangle function (x) has a unit height and a unit width centered at the origin.
Expanded by
The denominator of the argument of (x/) is the width 23 of the pulse.
Interpolation Sinc Function
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L'Hpital's Rule
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L'Hpital's Rule
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For a sinc(3/7) the first zero occurs at = 7/3. This is because the argument 3/7= when =7/3. =7/3
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Example 3.2
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Example 3.2
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Bandwidth of rect(t/)
Most of the signal Energy of the spectrum of the rect function is in the lower frequency components. components Signal Bandwidth: is the difference between the highest (significant) frequency and the lowest (significant) frequency in the signal spectrum. Much of the spectrum of the rect function is concentrated within the first lobe (from w=0 to w=2/. A rough estimate of the bandwidth of a rectangular pulse of width is 2/ rad/sec or 1/ Hz. 30
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Bandwidth of rect(t/)
To compute the bandwidth, one must consider the spectrum only for positive values of w. The trigonometric spectrum positive frequencies. exists only for
The negative frequencies occur because we use exponential spectra for mathematical convenience. Each sinusoid coswnt appears of two exponential compnents with frequencies wn and w-n. In reality there is only frequency which is the wn.
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one component
of
Example 3.3
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The function [sin(x)]/x is denoted by sinc(x). It is also known as the filtering or interpolating function.
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Example 3.4
or
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Example 3.5
We also have:
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Example 3.6
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Example 3.6
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Example 3.7
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Example 3.7
The transform of sgn (t) can be obtained by considering the sgn as a sum of two exponentials (in the limit a goes to zero) zero).
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Some Properties of the Fourier Transform
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Time Frequency Duality
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Symmetry Property
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Example 3.8
Apply the symmetry property to the pair that are shown below
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Example 3.8
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Example 3.8
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Example 3.8
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Scaling Property
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Scaling Property
The scaling property states that the time compression of a signal results in its spectral expansion, and time expansion of the signal results in its spectral compression.
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Example 3.9
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Example 3.9
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Time Shifting Property
Time delay in a signal causes a linear phase shift in its 51 spectrum.
Example 3.10
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Frequency Shifting Property
Multiplication of a signal shifts its spectrum by w=wo.

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by
factor
of
ejwot
Scaling Property
Changing wo to wo yields
ejwot is not a real function that can be generated. In practice frequency shift multiplying g(t) by a sinusoid as: is achieved by
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Scaling Property
Multiplication of a sinusoid cos (wot) by g(t) amounts to modulating the sinusoid amplitude. This type of modulation is called amplitude modulation. cos (wot) is called the carrier. The signal g(t) is called the modulating signal. g(t) cos (wot) is called the modulated signal signal. To sketch g(t) cos (wot)
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Amplitude Modulation
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Example 3.12
The Fourier transform of g(t) is
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Example 3.12
The Fourier transform of g(t) is
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Example 3.12
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Bandpass Signals
If gc(t) and gs(t) are low-pass signals, each with a bandwidth of B Hz or 2B rad/sec, then the signals gc(t) cos wot and gs(t) sin wot are both bandpass signals occupying the same band, and each having a bandwidth of 4B rad/sec. A linear combination of these signals will also be a bandpass signal occupying the same band as that of either signal and with the same bandwidth 4B g rad/sec A general bandpass signal can be expressed as
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Bandpass Signals
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Bandpass Signals
Slowly varying envelope Slowly varying phase

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Example 3.13
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Example 3.13
From Example 2.12
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Convolution
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Convolution
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Bandwidth of the Product of two Signals

If g1(t) and g2(t) have bandwidths B1 and B2 Hz, respectively. Then the bandwidth of g1(t) g2(t) is B1 + B2 Hz. Consequently, the bandwidth of g(t) is B Hz, then the bandwidth of g2(t) is 2B Hz, and the bandwidth f (t) is B Hz. b d id h of gn( ) i nB H
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Example 3.14
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Example 3.14
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Example 3.14
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Time Differentiation and Time Integration
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Example 3.15
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Example 3.15
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Example 3.15
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Signal Transmission Through a Linear System
For a linear, time invariant, continuous-time system the input-output relationship is given by
g(t) is the input, Y(t) is the output, and h(t) is the unit impulse response
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Signal Distortion During Transmission
The transmission of an input signal g(t) through a system changes it into the output signal y(t). G(w) and Y(w) are the spectra of the input and the output. H(w) is the spectral response of the system.
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Writing the equation in Polar form
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Distortionless Transmission
Transmission is said to be distortionless if the input and the output have identical wave shapes within a mu p multiplicative constant. n n . A delayed output that retains the input waveform is also considered distortionless. Thus, in distortionless transmission, the input g(t) and the output y(t) satisfy the condition
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Example 3.12
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Example 3.16
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Example 3.12
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Example 3.12
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Example 3.12
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Ideal and Practical Filters

The ideal low-pass filter allows all components below w=W rad/s to pass without distortion and suppresses all components above w=W.
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The ideal high-pass and characteristics are shown below. bandpass filter
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The ideal low-pass filter has a linear phase of slope -td, which results in a time delay of td seconds for all its input components of frequencies below W rad/s. If the input is a signal g(t) band-limited to W rad/s, the output y(t) is g(t) delayed by td, that is,
For this filter |H(w)| = rect (w/2W), and h(w)=wtd, so that

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The unit impulse response h(t) of this filter is:
The impulse response h(t) is not realizable. One practical approach to filter design is to cut off the tail of h(t) for t<0 The resulting causal impulse t<0. response h(t)
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Non-causal Causal
If td is sufficiently large, h(t) will be a close approximation of h(t), and the resulting filter ii (w) pp ( ), g ( ) will be a good approximation of an ideal filter. Theoretically a delay td = is needed to realize the ideal characteristics. But a delay td of three or four times /W will make h(t) a reasonably close 89 version of h(t-td).

The truncation operation [cutting the tail of h(t) to make it causal], however, creates some unsuspected problems of spectral spread and leakage This can be partly corrected by truncating h(t) gradually (rather than abruptly) using a tapered window function. Practical (realizable) filter characteristics gradual, without jump discontinuities in amplitude response |H(w)|.
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are the
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Butterworth Filters
The well-known Butterworth filters, for example, have amplitude response
The amplitude response approaches an ideal low-pass behavior as n. the Butterworth filter (half-power -3dB or a ratio of 1/2=0.707) bandwidth is B Hz. Also called cutoff frequency.
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Comparison of Butterworth Filter (n=4) with an ideal Filter
As n, the amplitude response approaches ideal, but the corresponding phase response is badly distorted in the vicinity of the cutoff frequency B Hz. A certain trade-off exists 92 between ideal magnitude and ideal phase characteristics.
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3.6 Signal Distortion over a Communication Channel

A signal transmitted over a channel is distorted because: of various channel imperfections. 1. Linear Distortion Signal distortion can be caused over a linear timeinvariant channel by nonideal characteristics of either the magnitude, the phase, or both. If a pulse g(t) l (t) is transmitted i t itt d through such a th h h channel. Spreading, or dispersion, of the pulse will occur if either the amplitude response or the phase response, or both, are nonideal.
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Linear Distortion
A distortionless channel multiplies each component by h b the same f factor and d l d delays each component b h by the same amount of time. If the amplitude response of the channel is not ideal [that is, |H(w)| is not equal to a constant], then the pulse will spread out (see the following example). example) The same thing happens if the channel characteristic is not ideal [h (w) -wtd].
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phase
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Linear Distortion
Dispersion of the pulse is undesirable in a TDM system, because pulse spreading causes interference with a neighboring pulse and consequently with a neighboring channel (crosstalk). For an FDM system, this type of distortion causes distortion (dispersion) in each multiplexed signal, but no interference occurs with a neighboring channel channel. This is because in FDM, each of the multiplexed signals occupies a band not occupied by any other signal.
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Example 3.17
A low-pass filter transfer function H(w) is given by
A pulse g(t) band-limited to B Hz is applied at the input of this filter. Find the output y(t).
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This filter has ideal phase and nonideal magnitude characteristics.
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The output is actually g(t) + (k/2)[g(t-T)+g(t+T)] delayed by td. It consists of g(t) and its echoes shifted by td. The dispersion of the pulse caused by its echoes is evident from the figure below.
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2. Distortion Caused by Channel Nonlinearities

We shall consider a simple case of a memoriless nonlinear channel where the input g and the output y are related by some nonlinear equation,
The right-hand side of this equation expanded in a McLaurin's series as
can
be
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If the bandwidth of g(t) is B Hz, then the bandwidth of gk(t) is kB Hz. Then, the bandwidth of y(t) is kB Hz Hz. The output spectrum spreads well beyond the input spectrum, and the output signal contains new frequency components not contained in the input signal. If a signal is transmitted over a nonlinear channel channel, the nonlinearity not only distorts the signal, but also causes interference with other signals on the channel because of its spectral dispersion (spreading) which will cause a serious interference 100 problem in FDM systems (but not in TDM systems).
Distortion Caused by Channel Nonlinearities
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Example 3.18
The input x(t) and the output y(t) of a certain nonlinear channel are related as 1. Find the output signal y(t) and its spectrum Y(w) if the input signal is x(t)=(1000/) sinc(1000t). 2. Verify that the bandwidth of the output signal is twice that of the input signal This is the result signal. of signal squaring. 3. Can the signal x(t) be recovered distortion) from the output y(t)?
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(without
Example 3.18
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Example 3.18
Observe that 0.316 sinc2(1000t) is the unwanted (distortion) term in the received signal.
Input signal spectrum X( ) t X(w)
Spectrum of the distortion term di t ti t
Received signal spectrum Y(w)

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Example 3.18
We make the following observations: 1. 1 The bandwidth of the received signal y(t) is twice that of the input signal x (t) (because of signal squaring).
Input signal spectrum X(w)

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Example 3.18
2. The received signal contains the input signal x(t) plus an unwanted signal (1000/) sinc2(1000t). Note that the desired signal and the distortion signal spectra overlap, and it is impossible to recover the signal x(t) from the received signal y(t) without some distortion.

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Example 3.18
3. We can reduce the distortion by passing the received signal through a low-pass filter of bandwidth I000 rad/s. Observe that the output of this filter is the desired input signal x(t) with some residual distortion.

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Spectrum of the Received signal after the lowpass filtering.
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Example 3.18
4. We have an additional problem of interference with other signals if the input signal x (t) is frequency-division multiplexed along with several other signals on this channel. This means that several signals occupying nonoverlapping frequency bands are transmitted simultaneously on the same channel. Spreading of the spectrum X(w) outside its original band of 1000 rad/s will interfere with the signal in the band of 1000 to 2000 rad/s. Thus, in addition to the distortion of x(t), we also have an interference with the neighboring band.
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Example 3.18
5. If x(t) were a digital signal consisting of a pulse train, each pulse would be distorted, but there would be no interference with the neighboring pulses. Moreover even with distorted pulses, data can be received without loss because digital communication can withstand considerable pulse distortion without loss of information. Thus, Thus if this channel were used to transmit a TDM signal consisting of two interleaved pulse trains, the data in the two trains would be recovered at the receiver.
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Example 3.18
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3. Distortion Caused by Multipath Effects

A multipath transmission takes place when a transmitted signal arrives at the receiver by two or more paths of different delays. In radio links, the signal can be received by direct path between the transmitting and the receiving antennas and also by reflections from other objects, such as hills, buildings, and so on. In this case the transmission channel can be represented as several channels in parallel, each with a different relative attenuation and a different time delay.
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Distortion Caused by Multipath Effects

For multipath signal propagation
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Let us consider the case of only two paths: one with a unity gain and a delay td, and the other with a gain and a delay td+t, as shown below.
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The transfer functions of the two paths are given by e-jwtd and e-jw(td+t), respectively. The overall transfer function of such a channel is H(w), given by
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Both the magnitude and the phase characteristics of H(w) are periodic in with a period of 2/t.
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The multipath transmission, therefore, causes nonidealities in the magnitude and the phase characteristics of the channel and will cause linear distortion (pulse dispersion). If the gains of the two paths are very close, that is, 1, the signals received by the two paths can very nearly cancel each other at certain frequencies, y y q where their phases are rad apart (destructive interference).
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At frequencies where =n/t (n odd), cos t=-1, q ( ), , and lH(w)l0 when 1. These frequencies are the multipath null frequencies. At frequencies =n/t (n even), the two signals interfere constructively to enhance the gain. Such channels cause frequency-selective fading of transmitted signals Such distortion can be partly signals. corrected by using the tapped delay-line equalizer.
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4. Fading Channels
Thus far, the channel characteristics were assumed , to be constant with time. In practice, we encounter channels whose transmission characteristics vary with time. These include troposcatter channels and channels using the ionosphere for radio reflection to achieve longdistance communication. The time variations of the channel properties arise because of semiperiodic and random changes in the propagation characteristics of the medium.
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Fading Channels
The reflection properties of the ionosphere, for example, example are related to meteorological conditions that change seasonally, daily, and even from hour to hour, much the same way as does the weather. Periods of sudden storms also occur. Hence, the effective channel transfer function varies semiperiodically and randomly, causing random attenuation of the signal This phenomenon is known signal. as fading. One way to reduce the effects of fading is to use automatic gain control (AGC).
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Fading Channels
Fading may be strongly frequency dependent where different frequency components are affected unequally. Such fading is known as frequencyselective fading and can cause serious problems in communication. Multipath propagation can cause frequency-selective fading.
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Signal Energy and Energy Spectral Density

The energy Eg of a signal g(t) is defined as the area under lg(t)l2 We can also determine the signal energy from its Fourier transform G(w) through Parseval's theorem. Signal energy can be related to the signal spectrum G(w) by:
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Example 3.19
Verify Parsevals theorem for the signal g(t)=e-atu(t) (a>0).
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Signal Energy and Energy Spectral Density

The energy spectral density (ESD) g(t) is given by:
at (t) is: The Th ESD of th signal g(t)=e-atu(t) i : f the i l (t)
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Essential Bandwidth of a Signal

Most of the signal energy is contained within a certain band of B Hz. Therefore, we can supress the signal spectrum beyond B Hz with little effect on the signal shape and energy. The bandwidth B is called the essentail bandwidth of the signal. Supression of all the spectral components of g(t) beyond the essential bandwidth results in a signal g^(t). If we use 95% criterion for the essential bandwidth, the energy of the error g(t)-g^(t) is 5% of Eg.
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Example 3.20
Estimate the essential bandwidth W rad/sec of the signal g(t)=e-atu(t) if the essential band is required to contain 95% of the signal energy.
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Example 3.20
The signal Eg is 1/2 times the area under the ESD, which is found to be 1/2a. Let W rad/sec be the essential bandwidth, which contains 95% of the total signal energy Eg. This means 1/2 ti 1/2 times th shaded area; the h d d
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Example 3.20
This means that all the remaining spectral components in the band from 12.706 to contribute only 5% of the signal energy.
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Example 3.21
Estimate the essential bandwidth of the a rectangular pulse g(t)=rect(t/T), where the essential bandwidth is to contain at least 90% of the pulse energy.
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Example 3.21
The energy EW within the band from 0 to W rad/sec gy is given by:
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Example 3.21
Setting wT=x in this integral so that dw=(1/T) dx;
Eg=T, we have;
Note that 90.28% of the total energy of the pulse g(t) is contained within the band W=2/T rad/sec or B=1/T Hz.
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Energy of Modulated Signals

Let g(t) be a baseband signal band-limited to b-Hz. The amplitude modulation (t) is:
The ESD of the modulated signal is
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Energy of Modulated Signals
The modulation shifts the ESD of g(t) by o. The area under (w) is half the area under g(w)
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Time Autocorrelation Function and the Energy Spectral Density

For a real signal g(t), the autocorrelation function g() is given by:
Setting x=t+
In this equation x is a dummy variable and could be replaced by t.
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This shows that for a real g(t), the autocorrelation function is an even function of We now show that the ESD g()=lG()l2 is the Fourier transform of the autocorrelation function g()
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The autocorrelation g() is the convolution of g() with g(-)
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Example 3.22
Find the time autocorrelation function of the signal g(t)=e-atu(t), and from it determine the ESD of g(t).
This is valid for positive . We can perform a similar procedure for negative . However, for real g(t), g() is an even function of . g , g( ), ( )
ESD of g() is the Fourier transform of g()

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Example 3.22
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ESD of the Input and the Output
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3.8 Signals Power and Power Spectral Density

Defining a truncated signal gT(t) as
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Power Spectral Density (PSD)

If the signal g(t) is a power signal, then its power is finite, and the truncated signal gT(t) is an energy signal as long as T is finite. From Parsevals Theorem
Pg, the power of g(t), is given by
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Power Spectral Density (PSD)

The power spectral density (PSD) Sg(w) is defined as:
The power spectral density (PSD) Sg(w) is defined as:
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Time Autocorrelation Function of Power Signals

The time autocorrelation function Rg() of a real power signal g(t) is defined as:
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The PSD Sg(w) is the Fourier transform of Rg()
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The concept and relationships for signal power are parallel to those for signal energy.
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Signal Power is its Mean Square Value

The signal power is the time average or mean of its squared value. In other words Pg is the mean square of g(t). A wavy overline is used to denote a time average.
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Interpretation of Power Spectral Density

The PSD Sg(w) represents the power per unit bandwidth (in hertz) of the spectral components at the frequency w. The power contributed by the spectral components within the band w1 to w2 is given by:
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Autocorrelation Method: a Powerful Tool

The Fourier transform is available only for deterministic signals, which can be described as functions of time. The random message signals that occur in communication problems, can not be described as functions of time, and its impossible to find their Fourier transforms. The autocorrelation function for such signals can be determined from their statistical information. This allows us to determine the PSD (the spectral information) of such a signal.
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Example 3.23
A random binary pulse train g(t) is shown below. The pulse width is Tb/2, and one binary digit is transmitted every Tb seconds. A binary 1 is transmitted by the positive pulse, and a binary 0 is transmitted by the negative pulse. The two symbols are equally likely and occur randomly. We shall determine the autocorrelation function, the PSD, and the essential bandwidth of this signal.
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Example 3.23
We cannot describe this signal as a function of time because the precise waveform is not known due to its random nature. We do know its behavior in terms of the averages (the statistical information). The autocorrelation function, being an average parameter (time average) of the signal, is determinable from the given statistical (average) information.
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Example 3.23
The figure below shows g(t) by solid lines and g(t-) by dashed lines. To determine the integrand on the right-hand side of the autocorrelation equation, we multiply g(t) with g(t-), find the area under the product g(t)g(t-), and divide it by the averaging interval T. Let there be N bits (pulses) during this interval T so that T = NTb, and as T , N .
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Example 3.23
Let us first consider the case of <Tb/2. In this case there is an overlap (shown by the shaded region) b t i ) between each pulse of g(t) and th t of g(th l f (t) d that f (t ). The area under the product g(t)g(t-) is Tb/2- for each pulse. Since there are N pulses during the averaging interval, then the total area under g(t)g(t) is N(Tb/2-),
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Example 3.23
Because Rg() is an even function of ,
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Example 3.23
As we increase beyond Tb/2, there will be overlap between each pulse and its immediate neighbor. The two overlapping pulses are equally likely to be of the same polarity or of opposite polarity. Their product is equally likely to be 1 or -1 over the overlapping interval. On the average, half the pulse products will be 1 (positive-positive ( iti iti or negative-negative ti ti pulse l combinations), and the remaining half pulse products will be -1 (positive-negative or negative-positive combinations).
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Example 3.23
Consequently, the area under g(t)g(t- ) will be zero when averaged over an infinitely large time (T ),
The autocorrelation function in this case is the triangle function 0.5(t-Tb). The PSD is the Fourier transform of 0.5(t-Tb), which is found in Example 3.15 as
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Example 3.23
The PSD is the square of the sinc function shown below. From the result in Example 3.21, we conclude that the 90 28% of the area of this spectrum is 90.28% contained within the band from 0 to 4/Tb rad/s, or from 0 to 2/Tb Hz. Thus, the essential bandwidth may be taken as 2/Tb Hz (assuming a 90% power criterion). This example illustrates dramatically how the p y autocorrelation function can be used to obtain the spectral information of a (random) signal where conventional means of obtaining the Fourier spectrum are not usable.
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Input and Output Power Spectral Densities

Because the PSD is a time average of ESDs, the relationship between the input and output signal PSDs of a linear time-invariant (LTI) system is similar to that of ESDs. If g(t) and y(t) are the input and output signals of an LTI system with transfer function H(w), then
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Example 3.24
A noise signal ni(t) with PSD Sni(w)=K is applied at the input of an ideal differentiator. Determine the PSD and the power of the output noise signal n0 (t).
The transfer function of an ideal differentiator is H(w)=jw. If the noise at the d ( ) f h h demodulator output is d l n0(t),
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Example 3.24
The output PSD Sno(w) is parabolic, as shown below. The output noise power N0 is l/2 times the area under the output PSD. Therefore,
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6/16/2012
PSD of Modulated Signals

For a power signal g(t), if
Then the PSD S (w) of the modulated signal (t) is given by Thus, Thus modulation shifts the PSD of g(t) by w0 The power of (t) is half the power of g(t), that is,
159
Homework #3
Solve the following problems: Due to one week from today 3.1-8 3.3-5 3.3-6 3.3-7 3.3-10 3.7-5 3.8-2
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Chapter 3 Analysis and Transmission of Signals Students

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Chapter 3 Analysis and Transmission of Signals Students

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6/16/2012

EE 23353 Analog Communications

Dr. Rami A. Wahsheh

Chapter 3: Analysis and Transmission of Signals

3.1 Aperiodic Signal Representation by Fourier Integral

3.1 Aperiodic Signal Representation by Fourier Integral

Third: Let the period

3.1 Aperiodic Signal Representation by Fourier Integral

3.1 Aperiodic Signal Representation by Fourier Integral

3.1 Aperiodic Signal Representation by Fourier Integral

3.1 Aperiodic Signal Representation by Fourier Integral

Spectral components are spaced p p p at zero (infinitesimal) interval

gTo(t) can be expressed as a sum of everlasting exponentials of frequencies

Conjugate Symmetry Property

Linearity of the Fourier Transform

3.2 Transforms of Some Useful Functions

The denominator of the argument of rect(x/) is the 22 width of the pulse.

Unit Triangle Function

The denominator of the argument of (x/) is the width 23 of the pulse.

Interpolation Sinc Function

Interpolation Sinc Function

Interpolation Sinc Function

Some Properties of the Fourier Transform

Time Frequency Duality

Time Shifting Property

Time delay in a signal causes a linear phase shift in its 51 spectrum.

Frequency Shifting Property

Multiplication of a signal shifts its spectrum by w=wo.

The Fourier transform of g(t) is

Slowly varying envelope Slowly varying phase

From Example 2.12

Bandwidth of the Product of two Signals

Time Differentiation and Time Integration

Signal Transmission Through a Linear System

Signal Distortion During Transmission

Signal Distortion During Transmission

Writing the equation in Polar form

Signal Distortion During Transmission

Ideal and Practical Filters

Ideal and Practical Filters

Signal Distortion During Transmission

For this filter |H(w)| = rect (w/2W), and h(w)=wtd, so that

Ideal and Practical Filters

Ideal and Practical Filters

Ideal and Practical Filters

Comparison of Butterworth Filter (n=4) with an ideal Filter

3.6 Signal Distortion over a Communication Channel

This filter has ideal phase and nonideal magnitude characteristics.

2. Distortion Caused by Channel Nonlinearities

The right-hand side of this equation expanded in a McLaurin's series as

Distortion Caused by Channel Nonlinearities

Input signal spectrum X( ) t X(w)

Spectrum of the distortion term di t ti t

Received signal spectrum Y(w)

Input signal spectrum X(w)

Received signal spectrum Y(w)

Received signal spectrum Y(w)

Received signal spectrum Y(w)

Spectrum of the Received signal after the lowpass filtering.

3. Distortion Caused by Multipath Effects

Distortion Caused by Multipath Effects

Distortion Caused by Multipath Effects