Baseband Modulation

Chapter 3 Baseband Pulse Transmission and
Digital Modulation Techniques

3.1 Baseband Pulse Transmission
Line Coding
Eye Patterns
Intersymbol Interference
Nyquist Waveshaping
Adaptive Equalization
3.2 Transmission Over Band-Pass Channel

Amplitude-Shift Keying (ASK)
Frequency-Shift Keying (FSK)
Phase-Shift Keying (PSK)
Differential Phase Shift Keying (DPSK)
M-ary Modulation
Quadrature Amplitude Modulation
Minimum-Shift-Keying 1
Line Coding
How do we mathematically represent the waveform for a
digital signal?
Binary 1’s and 0’s, such as in PCM signalling, may be
represented in various serial-bit signalling formats called line
codes.
Line codes are also used for controlling and shaping the
power spectral density of a digital communication signal so
that the spectrum of the transmitted signal matches the
spectral characteristics of a baseband or equivalent lowpass
channel.
Choosing the appropriate line code can minimise the effect of
channel distortion and noise in transmission.
There are two major categories: return-to-zero (RZ) and
nonreturn-to-zero (NRZ).
2
With RZ coding, the waveform returns to a zero-volt level for
Binary Line Codes
Unipolar Signaling : binary 1 is represented by a high

level (+A volts) and a binary 0 by a zero level; also called
on-off keying.
Polar Signaling : binary 1’s and 0’s are represented by
equal positive and negative levels.
Bipolar Signaling : binary 1’s are represented by
alternately positive or negative values. The binary 0 is
represented by a zero level, also called alternate mark
inversion (AMI) signaling.
Manchester Signaling : each binary 1 is represented by a
positive half-bit period pulse followed by a negative half-bit
period pulse while a binary 0 is represented by a negative
half-bit period pulse followed by a positive half-bit period
3
pulse. Also called split-phase encoding.
4
Desirable Properties of Line Codes
Self-synchronisation: There is enough timing information
built into the code so that bit synchronizers can be
designed to extract the timing or clock signal.
Low probability of bit error: Receivers can be designed
that will recover the binary data with a low probability of bit
error when the signal is corrupted by noise or ISI.
A spectrum that is suitable for the channel
Transmission bandwidth: This should be as small as
possible
Error detection capability: It should be possible to
implement channel coding techniques.
Transparency: The data protocol and line code are
designed so that every possible sequence of data can be
transmitted and received. 5
Eye Patterns
An eye pattern or eye diagram is a simple and convenient
tool for studying the effects of intersymbol interference and
other channel impairments in digital signal transmission.
The eye pattern is defined as the synchronized superposition
of all possible realisations of the signal of interest viewed
within a particular signalling interval.
To construct an eye pattern, we plot the received signal
against time on a fixed-interval axis, at the end of the fixed
time interval, wrap around to the beginning of the time axis.
Thus the diagram consists of many overlapping curves.
The eye pattern derives its name because it resembles the
human eye for binary waves.
It provides an excellent way of assessing the quality of the
received line code and the ability of the receiver to combat6 bit
errors.
7
Best
sampling
Slope = sensitivity time Distortion at
to timing error sampling time
Noise margin
Time interval over which the Distortion of

received signal can be sampled zero-crossings
Interpretation of the eye pattern
8
The eye pattern provides the following information:
The timing error allowed on the sampler at the receiver is given by
the width inside the eye, called the eye opening. Of course, the
preferred time for sampling is at the point where the vertical
opening of the eye is the largest.
The sensitivity to timing error is given by the slope of the open
eye(evaluated at, or near, the zero-crossing point).
The noise margin of the system is given by the height of the eye
opening.
In the presence of ISI and channel noise, traces from the

upper portion of the eye pattern cross traces from the lower
portion.
This causes the eye to close, thereby reducing the margin for
additive noise to cause errors. 9
Eye Diagram: low noise and ISI-free chanel Eye Diagram: noisy and ISI chanel
1.5 2
1.5
1
1
0.5
0.5
Amplitude
Amplitude
0 0
-0.5 -0.5
-1
-1
-1.5
-1.5
Time -2
Time
Eye pattern for binary signalling
10
In the case of an multilevel system, the eye pattern
contains (L-1) eye openings stacked up vertically one on
the other, where L is the number of discrete amplitude
levels used to construct the transmitted signal.
Eye Diagram: low noise and ISI-free chanel Eye Diagram: noisy and ISI chanel
1
1
0.5
0.5
Amplitude
Amplitude
0
0
-0.5
-0.5
-1
-1
Time
Time
Eye patterns for L= 4 signal
11
For PSK (phase-shift keying) modulation, it is customary
to display the “eye pattern” as a two-dimensional scatter
diagram illustrating the sampled values that represent
the decision variables at the sampling instants.
Noiseless and ISI-free channel Noisy and ISI channel

1.5 1.5
1 1
0.5 0.5
Quadrature
Quadrature
0 0
-0.5 -0.5
-1 -1
-1.5 -1.5
-1.5 -1 -0.5 0 0.5 1 1.5 -1.5 -1 -0.5 0 0.5 1 1.5
In-Phase In-Phase
Scatter plot for an 8-PSK signal

12
Regenerative Repeaters
When a line code digital signal (such as PCM) is transmitted
over a hardwire channel (such as a twisted-pair telephone
line), it is attenuated, filtered, and corrupted by noise.
Consequently, for long lines, the data cannot be recovered at
the receiving end unless repeaters are placed in cascade
along the line and at the receiver.
These repeaters amplify and “clean up” the signal periodically.

For analog signals, only linear amplifiers can be used (since relative
amplitude values would need to be preserved) and the in-band
distortion would accumulate from linear repeater to linear repeater.
This is one of the disadvantages of analog signaling. 13
With digital signaling, nonlinear processing can be used to
regenerate a “noise-free” digital signal. This type of nonlinear
processing is called a regenerative repeater.
The amplifying filter increases the amplitude of the low-level input

signal so that is compatible with the remaining circuitry and filters
the signal so as to minimize the effects of channel noise and ISI.
The bit synchronizer generates a clocking signal.
14
For each clock pulse, the sample and hold circuit produces the
sample value that is held for 1-bit interval until the next clock pulse.
The comparator produces a high-level output only when the sample
value is larger than the threshold level, VT . VT is selected to be one-
half the expected peak-to-peak variation of the sample values.
If the channel noise and ISI is negligible, the comparator output will
be high only when there is a binary 1 (i.e., a high level) on the
corrupted unipolar NRZ line code at the input to the repeater.
15
The comparator (a threshold device) acts as a decision-making
device.
Thus, the unipolar NRZ line code is regenerated “noise free” except
for bit errors that are caused when the input noise and ISI alter the
sample values sufficiently so that the sample values occur on the
wrong side of VT .
The bit error rate is influenced by the SNR at the input to the
repeater, the filter that is used, and the value of VT that is selected.
16
In long-distance digital communication systems, many repeaters
may be used in cascade.
Of course, the spacing between the repeaters is governed by the path
loss in the transmission medium and the amount of noise that is
added.
A repeater is required when the SNR at a point along the channel
becomes lower than the value that is needed to maintain the overall
probability-of-bit-error specification. 17
Bit Synchronization
Synchronization signals are clock-type signals that are
needed in a receiver for detection of the data from the
corrupted input signal.
These clock signals have a precise frequency and phase
relationship with respect to the received input signal.
Digital communications usually require three types of sync
signals :
1. Bit sync, to distinguish one bit interval from another;
2. Frame sync, to distinguish groups of data in regard to time-division multiplexing;
3. Carrier sync, for passband signalling with coherent detection.
Systems are designed so that the sync is derived either

directly from the corrupted signal or from a separate channel
that is used only to transmit the sync information.
Bit synchronizers that derive the sync directly from the
18
corrupted signal are more desirable.
PSD for polar NRZ and
unipolar RZ line codes
19
The complexity of the bit synchronizer circuit depends on the
sync properties of the line code.
Example : A unipolar RZ code with a sufficient number of
alternating binary 1’s and 0’s is almost trivial, since the PSD
of that code has a delta function at the bit rate, f = R.
The bit sync clock signal can be obtained by passing the
received unipolar RZ waveform through a narrowband
bandpass filter that is tuned to f0 = R = 1/Tb.
For a polar NRZ line code, the bit synchronizer is slightly
more complicated. The filtered polar NRZ waveform is
converted to a unipolar RZ waveform by using a square-law
(or, alternatively, a full-wave rectifier) circuit.
The clock signal is then recovered using a filter or a PLL,
since the unipolar RZ has a delta function at f = R.
20
21
Unipolar, polar, and bipolar bit synchronizers will work only
when there are a sufficient number of alternating 1’s and 0’s
in the data.
The loss of synchronization because of long strings of all 1’s
or all 0’s can be prevented by adopting one of two possible
alternatives.
One alternative is to use bit interleaving (i.e., scrambling).
The other alternative is to use a completely different type of
line code that does not require alternating data for bit
synchronization.
For example, Manchester NRZ encoding can be used, but it
will require a channel with twice the bandwidth of that
needed for a polar NRZ code.
22
Intersymbol Interference
The absolute bandwidth of rectangular multilevel pulses is
infinity.
23
Digital data have a broad spectrum with a significant low-
frequency content.
Digital baseband transmission requires a lowpass channel
with a bandwidth sufficient to accommodate the essential
frequency content of the data stream.
If these pulses are filtered improperly as they pass through a
communication system, they will spread in time, and the
pulse for each symbol may be smeared into adjacent time
slots and cause intersymbol interference (ISI).
ISI is a major source of bit errors in the receiver. To correct it,
control has to be exercised over the pulse shape in the
overall system. Another approach to mitigate ISI is to equip
the receiver with an equaliser.
But…how does ISI arise???
24
Coherence bandwidth is a measure of the range of
frequencies over which all spectral components with
approximately equal gain and linear phase.
x(t ) h(t , τ) y (t )
x(t ) h(t , τ) y (t )
τ << Ts
t t t
0 Ts 0τ 0 Ts + τ
X(f ) H( f ) Y( f )
f f f
fc fc fc
25
Flat fading channel characteristics
x(t ) h ( t , τ) y (t )
x(t ) h(t , τ) y (t )
t t t
0 Ts 0 τ 0 Ts Ts + τ
X( f ) H( f ) Y( f )
f f f
fc fc fc
Frequency selective fading channel characteristics

ISI is generated! 26
Can we restrict the bandwidth without introducing ISI?
Of course, with restricted bandwidth, the pulses would have
rounded tops (instead of flat tops).
27
Flat-topped multilevel signal at the input is:
win (t ) = ∑ an h(t − nTs )
n
where h(t)= ∏(t/Ts) and an may take on any of the allowed L

multilevels (L = 2 for binary signaling).
win (t ) = ∑ an h(t ) ∗ δ (t − nTs )
n
⎡ ⎤
= ⎢∑ anδ (t − nTs )⎥ ∗ h(t )
⎣ n ⎦
28
The output of the linear system would be just the input
impulse train convolved with the equivalent impulse response
of the overall system.
⎡ ⎤
wout (t ) = ⎢∑ anδ (t − nTs )⎥ ∗ he (t )
⎣ n ⎦
where the equivalent impulse response is
he (t ) = h(t ) ∗ hT (t ) ∗ hC (t ) ∗ hR (t )
The equivalent system transfer function is

H e ( f ) = H ( f )H T ( f )H C ( f )H R ( f )
where ⎡ ⎛ t ⎞⎤ ⎛ sin πTs f ⎞

H ( f ) = F ⎢∏⎜⎜ ⎟⎟⎥ = Ts ⎜⎜ ⎟⎟
⎣ ⎝ Ts ⎠⎦ ⎝ πTs f ⎠
when flat-top pulses are used at the input of transmitter.
29
The receiving filter is given by
He ( f )
HR( f ) =
H ( f )H T ( f )H C ( f )
where He(f) is the overall filtering characteristic.
When He(f) is chosen to minimize the ISI, HR(f) is the transfer
function of an equalizing filter.
The equalizing filter characteristic depends on HC(f), the
channel frequency response, as well as on the required He(f).
When the channel consists of dial-up telephone lines, the
channel transfer function changes from call to call and the
equalizing filter may need to be an adaptive filter.
In this case, the equalizing filter adjusts itself to minimize the
ISI.
A test bit pattern that is used to adapt the filter electronically
for the maximum eye opening (i.e., minimum ISI).
30
Such sequences are called learning or training sequences.
Nyquist’s First Method (Zero ISI)
Nyquist’s first method for eliminating ISI is to use an
equivalent transfer function, He(f), such that the impulse
response satisfies the condition.
⎧C , k = 0
he (kTs + τ ) = ⎨
⎩ 0, k ≠ 0
where k is an integer, Ts is the symbol (sample) clocking
period, τ is the offset in the receiver sampling clock times
compared with the clock times of the input symbols, and C is a
nonzero constant.
It means that for a single flat-top pulse of level a present at the input
to transmitting filter at t = 0, the received pulse would be aC at t = τ
but would not cause interference at other sampling times because
he(kTs + τ) = 0 for k ≠ 0.
31
If we choose a (sin x)/x function for he(t). Let τ =0 and choose
sin πf s t
he (t ) =
πf s t
where fs =1/Ts is called the Nyquist rate.
The sinc impulse response satisfies Nyquist’s first criterion
for zero ISI.
Consequently, if the transmit and receive filters are designed
so that the overall transfer function is
1 ⎛ f ⎞
H e ( f ) = ∏⎜⎜ ⎟⎟
fs ⎝ fs ⎠
there will be no ISI.
Absolute bandwidth of this transfer function is B = fs /2, the
optimum filtering to produce a minimum-bandwidth system.
It will allow signaling at a baud rate of D = 1/Ts = 2B pulses/s,
where B is the absolute bandwidth of the system. 32
(a) Ideal magnitude response. (b) Ideal basic pulse shape.
33
Binary sequence 1 0 1 1 0 1 0
0.8
0.6
0.4
0.2
Amplitude
-0.2
-0.4
-0.6
-0.8
-1
-2 -1 0 1 2 3 4 5 6 7 8 9 10
Time
A series of sinc pulses corresponding to the sequence 1011010

34
The sinc type of overall pulse shape has two practical
difficulties:
The overall amplitude transfer characteristic He(f) has to be
flat over –B < f < B and zero elsewhere. This is physically
unrealizable (i.e., the impulse response would be
noncausal and of infinite duration).
The synchronization of the clock in the decoding sampling
circuit has to be almost perfect, since the sinc pulse decays
only as 1/x and is zero in adjacent time slots only when t is
at the exact sampling instant. This inaccurate sync will
cause considerable ISI.
Because of these problems, we should consider other pulse shapes
that have a slightly wider bandwidth.
The idea is to find pulse shapes that go through zero at adjacent
sampling points and yet have an envelope that decays mush faster
than 1/x. One solution for the equivalent transfer function, which
has many desirable features, is the raised cosine-rolloff Nyquist
35
filter.
Raised Cosine-Rolloff Nyquist Filtering
The raised cosine-rolloff Nyquist filter has the transfer
function ⎧1, f < f1
⎪⎪ 1 ⎧ ⎡π ( f − f1 )⎤ ⎫
H e ( f ) = ⎨ ⎨1 + cos ⎢ ⎥ ⎬, f1 < f < B
⎪2 ⎩ ⎣ 2 fΔ ⎦⎭
⎪⎩0, f >B
where B is the absolute bandwidth and the parameters
fΔ = B − f0 and f1 = f 0 − f Δ
f0 is the 6-dB bandwidth of the filter and the roll-off factor is
defined to be f
r= Δ
f0
The corresponding impulse
response is ⎛ sin 2πf 0t ⎞ ⎡ cos 2πf Δ t ⎤
he (t ) = F [H e ( f )] = 2 f 0 ⎜⎜
−1
⎟⎟ ⎢ 2⎥
⎝ 2π f 0 t ⎠ ⎣1 − (4 f Δ t ) ⎦ 36
37
38
As the absolute bandwidth is increased (e.g., r = 0.5 or r =
1.0)The
1. : filtering requirements are relaxed, although h (t) is
e
still noncausal
2. The clock timing requirements are relaxed also, since
the envelope of the impulse response decays faster than
1/|t|(on the order of 1/|t|³ for large values of t).
To determine the baud rate that can be supported by the

raised cosine-rollof system, note that the data pulses can
inserted at t = n/2f0 where n ≠ 0. Therefore, baud rate, D =
1/Ts = 2f0.
In other words, the 6-dB bandwidth of the raised cosine-rolloff
filter, f0 is designed to be half the symbol (baud) rate.
The transmission bandwidth is defined by
B = f 0 + f Δ = f 0 + rf 0 = (1 + r ) f 0 39
However, the filter is noncausal. We could use a filter with
a linear phase characteristic He(f)e-jωTd, and there would be
on ISI if we delayed the clocking by Td sec, since the e-jωTd
factor is the transfer function of an ideal delay line.
h(t − Td ) ↔ H ( f )e − jωTd
Time delay
property
This would move the peak of the impulse response to the
right (along the time axis), and then the filter would be
approximately causal.
40
Adaptive Equalization
In a telecommunications environment, the channel is usually
time-varying. For example, in a switched telephone network, we
find that two factors contribute to the distribution of pulse
distortion on different link connections:
Differences in the transmission characteristics of the individual
links that may be switched together.
Differences in the number of links in a connection.
The result is that the telephone channel is random in the sense of

being one of an ensemble of possible physical realizations.
Consequently, the use of a fixed equalizer designed on the basis
of average channel characteristics may not adequately reduce
ISI. To realize the full transmission capability of a telephone
channel, there is need for adaptive equalization.
41
Adaptive equalizer adjusts itself continuously and automatically
by operating on the input signal.
Two types of adaptive equalization: pre-channel equalization
at the transmitter; post-channel equalization at the receiver.
The first approach requires a feedback channel, we consider only
adaptive equalization at the receiving end of the system.
This equalization can be achieved, prior to data transmission, by
training the filter with the guidance of a suitable training
sequence transmitted through the channel so as to adjust the
filter parameters to optimum values.
We study an adaptive equalizer based on the tapped-delay-line
filter, which is synchronous in the sense that the tap spacing of
the equalizer is the same as the symbol duration T of the
transmitted signal.
This equalizer is simple to implement and its performance is a
42
satisfactory.
Block diagram of adaptive equalizer
The output yn of the tapped-delay-line equalizer in response to the

input sequence {xn} is defined by the discrete convolution sum:
N
yn = ∑w x
k =− N
k n−k
Where wk is the weight at the kth tap. 43

The adaptation uses the least mean square (LMS) algorithm.
⎛ Old value ⎞ ⎛ Input signal ⎞
⎛ Updated value ⎞ ⎜ ⎟ ⎛ Step - size ⎞ ⎜ ⎟⎛ Error ⎞
⎜⎜ ⎟⎟ = ⎜ of kth tap - ⎟ + ⎜⎜ ⎟⎟⎜ applied to kth ⎟⎜⎜ ⎟⎟
⎝ of kth tap - weight ⎠ ⎜ weight ⎟ ⎝ parameter ⎠⎜ kth tap - weight ⎟⎝ signal ⎠
⎝ ⎠ ⎝ ⎠
wˆ k [n + 1] =
wˆ k [n] + μx[n − k ]e[n]
Signal-flow graph of the

LMS algorithm for the kth
tap weight
44
There are two modes of operation for an adaptive equalizer,
namely, the training mode and decision-directed mode.
During the training mode, a known sequence is transmitted and a

synchronized version of this signal is generated in the receiver,
where (after a time shift equal to the transmission delay) it is
applied to the adaptive equalizer as the desired response.
45
The tap-weights of the equalizer are thereby adjusted in
accordance with the LMS algorithm.
A training sequence commonly used in practice is the so-called
pseudo-noise (PN) sequence, which consists of a deterministic
sequence with noise-like characteristics.
When the training process is completed, the adaptive equalizer is
switched to its decision-directed mode by using its output directly
for adaptation. 46
In normal operation the decisions made by the receiver are correct
with high probability.
This means that the error estimates are correct most of the time,
thereby permitting the adaptive equalizer to operate satisfactorily.
Furthermore, an adaptive equalizer operating in a decision-
directed mode is able to track relatively slow variations in channel
characteristics.
The larger the step-size parameter μ the faster the tracking
capability of the adaptive equalizer.
However, a large step-size parameter μ may result in an
unacceptably high excess mean-square error (MSE).
We therefore find that in practice the choice of a suitable value
for the step-size parameter μ involves making a compromise
between fast tracking and reducing the excess mean-square error.
47
Convergence characteristics of the LMS
algorithm with different step sizes 48
Transmission Over Bandpass Channels
In baseband pulse transmission, a data stream represented in the
form of a discrete PAM signal is transmitted directly over a low-
pass channel.
An issue of particular concern in baseband pulse transmission is
that of pulse shaping designed to bring the ISI under control.
In digital bandpass transmission, on the other hand, the incoming
data stream is modulated onto a carrier (usually sinusoidal) with
fixed frequency limits imposed by a bandpass channel of interest.
The major issue of concern here is the optimum design of the
receiver so as to minimize the average probability of symbol error
in the presence of noise.
The communication channel used for passband data transmission
may be a microwave radio link, a satellite channel, or the like.
49
The modulation process making the transmission possible
involves switching the amplitude, frequency, or phase of a
sinusoidal carrier in accordance with the incoming data.
Thus there are three basic digital signaling schemes known as
Amplitude shift keying (ASK)
Frequency shift keying (FSK)
Phase shift keying (PSK)
which may be viewed as special cases of amplitude modulation,
frequency modulation, and phase modulation, respectively.
A distinctive feature of FSK and PSK signals is that they both
have a constant envelope.
This feature makes them impervious to amplitude non-linearity,
commonly encountered in microwave radio links and satellite
channels.
50
Therefore in practice, FSK and PSK signals are preferred to ASK
signals for digital passband transmission over nonlinear channels.
51
Amplitude-Shift Keying (ASK)
Assume a sequence of binary pulses as shown. The l's turn on the
carrier of amplitude A, the 0's turn it off.
The spectrum of the ASK signal will depend on the particular
binary sequence to be transmitted.
Call a particular sequence of l's and 0's f(t). Then the amplitude-
modulated signal is simply f (t ) = Af (t ) cos ω t
c c
52
The effect of multiplication by cosωct is simply to shift the
spectrum of the original binary signal (the baseband signal) up to
frequency ωc.
This is the double-sideband suppressed carrier (DSB-SC) AM
signal. It contains upper and lower side bands symmetrically
distributed about the carrier or center frequency ωc.
(a) Spectrum of modulating signal (b) Spectrum of AM wave 53

The shaped modulated signal will have the spectrum the baseband
signal shifted up to, and centered about, the carrier frequency.
This applies to any other types of pulse shaping.
Example with sinusoidal roll-off shaping

As an example, say that sinusoidal/raised cosine roll-off shaping
is used, either by shaping the baseband pulses, or by shaping the
high-frequency modulated pulses. 54
The spectrum of the modulated signal looks like the baseband
spectrum, shifted up to the carrier frequency fc hertz and with a
transmission bandwidth
1
BT = 2 B = (1 + r )
Ts
with r the roll-off factor.
The transmission bandwidth is

B = f 0 + f Δ = f 0 + rf 0 = (1 + r ) f 0 55
1
Ts =
2 f0
B = f 0 (1 + r )
1
= (1 + r )
2Ts
Therefore, the transmission bandwidth is

1
BT = 2 B = (1 + r )
Ts 56
The frequency shift of a signal f(t) due to multiplication by cosωct
is a general result for AM signals.
It is true for all modulating signals f(t) and not just for the binary
case we are in the process of considering.
As an example, let f(t) = cosωct, a single sine wave of frequency
ωc. Then, by trigonometry,
1 1
cos ω m t cos ω c t = cos(ω m + ω c )t + cos(ω m − ω c )t
2 2
The single-line spectral plot representing cosωct is thus replaced

by two lines, symmetrically arrayed about ωc.
Similarly, if f(t) is a finite sum of sine waves (periodic signals),
each sine wave is translated up in frequency by ωc.
57
For a binary train of alternating l's and 0's, resulting in a
periodically alternating ASK signal, the spectrum of this signal is
just the (sinx)/x line spectrum of a pulse of width T, periodic with
period 2T, translated up to frequency fc.
(a) periodic ASK signal, (b) Spectrum (positive frequency only)

58
Frequency-Shift Keying (FSK)
Binary FSK modulated signal is represented as:
f c (t ) = A cos ω1t ⎫ T T
⎬ − ≤ t ≤
f c (t ) = A cos ω 2 t ⎭ 2 2
A l corresponds to frequency f1, a zero to frequency f2.

Generally, f1 and f2 >> 1/T.
An alternative representation of the FSK wave consists of letting
f1 = f c − Δf , f 2 = f c + Δf ,
59
The FSK wave can also be written as:
T T
f c (t ) = A cos(ω c ± Δω)t , − ≤ t ≤
2 2
The frequency then deviates ±Δf about fc .
The quantity Δf is commonly called the frequency deviation.
The frequency spectrum of the FSK wave is difficult to obtain.
We consider one special case, which provides insight into the
spectral characteristics of more complex FM signals, and leads to
a good rule of thumb regarding FM bandwidths.
Assume that the binary message consists of an alternating
sequence of ls and 0s.
If the two frequencies are each multiples of the reciprocal of the
binary period T (i.e., f1 = m/T, f2 = n/T, m and n integers), and are
synchronized in phase. 60
Then this FSK wave is a periodic function:
Note, however, that this may also be visualized as the linear

superposition of two periodic ASK signals, one delayed T
seconds with respect to the other.
The spectrum is then the linear superposition of two spectra.
Specifically, it can be shown that the positive frequency spectrum
has the form
sin[(ω1 − ω n )T / 2] sin[(ω 2 − ω n )T / 2]
+ (−1) n
(ω1 − ω n )T / 2 (ω 2 − ω n )T / 2
61
Spectrum of periodic FSK wave (positive frequency only)
The bandwidth of this periodic FSK signal is then 2Δf + 2B, with
B the bandwidth of the baseband signal.
Two extreme cases are of interest:
1. Wideband FM
2. Narrow band FM
62
In wideband FM, Δf >> B, and the bandwidth approaches 2Δf.
Thus in wideband FM, the bandwidth is essentially just that
separation.
It is virtually independent of the bandwidth of the baseband
binary signal. This is distinctly different from the AM case.
In narrowband FM, Δf << B, and the bandwidth approaches 2B.
Here the bandwidth is determined by the baseband signal.
If the baseband signal is an arbitrary string of binary pulses, each
shaped according to sinusoidal shaping with a roll-off factor, r,
the approximate bandwidth of the corresponding FSK signal is
given by
BT ( FM ) = 2Δf + 2 B
1
with B= (1 + r )
2T
T being the baseband (or FSK) pulse width. 63
The exact shape of the FSK spectrum is difficult to calculate, but
its form would be roughly that shown below.
FSK spectrum, sinusoidal roll-off shaping (positive frequency only)
Note that the FM transmission bandwidth is generally much

greater than that for AM, which is always 2B, that is, twice the
baseband bandwidth.
Then why use FM? 64
It is this wideband property of FM that makes its performance
generally far superior to AM in a noisy environment. (shown
later)
This is analogous to the pulse modulation.
Encoding of pulse-amplitude-modulation (PAM) signals into
binary pulse code modulation (PCM) results in an expansion of
the system bandwidth, but the noise immunity increases
considerably.
A general characteristic of communication systems is that one can
generally improve system performance in the presence of noise
by encoding or modulating signals into equivalent wideband
forms.
Binary PCM and FM are examples of such wideband signals.
In FM analysis, the modulation index is defined as the ratio of
the frequency deviation Δf and base
Δf band bandwidth B:
β≡ 65
B
In terms of β the FM transmission bandwidth is:
BT ( FM ) = FM bandwidth = 2Δf + 2 B = 2 B (1 + β)
Carson’s rule:
98% of the total power is contained within the bandwidth BT(FM).
Narrowband FM systems correspond to β << 1.

Wideband FM systems correspond to β >> 1.
The modulation index β plays a very significant role in the
analysis of FSK systems.
66
Phase-Shift Keying (PSK)
Binary PSK signal using rectangular shaping is given as:
T T
f c (t ) = ± cos ω c t , − ≤ t ≤
2 2
Here a 1 in the baseband binary stream corresponds to positive
polarity, and a 0 to negative polarity.
The PSK signal thus corresponds essentially to a polar NRZ
binary stream, translated up in frequency.
The PSK signal has the same double-sideband characteristic as

67
ASK transmission.
Differential Phase-Shift Keying (DPSK)
When serial data are passed through many circuits along a
communication channel, the waveform is often unintentionally
inverted (data complemented).
For example, this can happen in twisted-pair transmission line
channel just by switching the two leads at a connection point when
a polar line code is used.
DPSK is a noncoherent form of PSK that overcomes this problem
while without requiring a coherent reference signal at the receiver.
Noncoherent receivers are easy and cheap to build, and hence are
widely used in wireless communications.
A DPSK transmitter combines two basic operations:
Differential encoding
Phase-shift keying
68
In DPSK, the input binary data stream is differentially encoded
into a new binary data stream.
The new data stream represents a 1 by a change in polarity and a 0
by no change in polarity. (⇒ Exclusive-OR operation)
This operation is made clear by the following example.
69
70
71
M-ary Modulation
The bandwidth required for transmitting a baseband digital
sequence could be reduced by using multilevel signaling:
combining successive binary pulses to form a longer pulse
requiring a correspondingly smaller bandwidth for transmission.
Specifically, with ideal Nyquist shaping 2 (symbols/s)/Hz can be
transmitted over the Nyquist bandwidth of B hertz. If a set of M =
2k symbols is used, with k the number of successive binary digits
combined to form the appropriate symbol to be transmitted, 2k
(bits/s)/Hz may be transmitted using the Nyquist band.
72
73
Multi-symbol systems include multi-phase, multi-amplitude, and
combined multi-phase/multi-amplitude signaling schemes.
These are commonly used in telephone, microwave, and satellite
data communications to achieve higher spectrum efficiency.
Multi-symbol signals are often called M-ary signals.
Consider a system in which two successive binary pulses are
combined, and the resultant set of four binary pairs, 00, 01, 10, 11,
is used to trigger a high-frequency sine wave of four possible
phases, one for each of the binary pairs.
The ith signal, of the four possible ones, can be written
T T
si (t ) = cos(ω c t + θ i ), i = 1, 2, 3, 4, − ≤ t ≤
2 2
Two possible choices for the four phase angles are:
π π 3π
θ i = 0,± , π θ i = ± ,±
2 4 4 74
In both cases the phases are spaced π/2 radians apart.
Signals of this type are called quaternary/quadrature PSK
(QPSK) signals. They are a special case of multi-PSK (MPSK)
signals. Binary PSK signals often labeled as BPSK.
In general, k successive binary pulses are stored up and one of M =
2k symbols is output.
If the binary rate is R bits/s, each binary pulse interval is 1/R
seconds long. The corresponding output symbol is then T = k/R
seconds long.
The QPSK signals may be represented in the following form:
T T
s i (t ) = a i cos ω c t + bi sin ω c t , − ≤t≤
2 2
where ai and bi are the in-phase and quadrature components,
respectively.
75
Transmission of this type is often called quadrature transmission,
with two carriers in phase quadrature to one another (cosωct and
sinωct) transmitted simultaneously over the same channel.
It is useful to represent the MPSK signals in a two-dimensional
constellation diagram by locating the various points (ai, bi).
The horizontal axis corresponding to the location of ai, is called the
in phase axis. The vertical axis, along which bi, is located, is called
the quadrature axis.
QPSK signal constellation 76

The in-phase (cosine) and quadrature (sine) representation of the
QPSK signals si(t) suggests a method to generate these signals:
Two successive binary input pulses are stored up, and the pair of numbers (ai,bi ),
taken every T = 2/R seconds, is used to modulate two quadrature carrier terms,
cosωct and sinωct respectively.
QPSK modulator 77
It is apparent that demodulation is carried out by using two
synchronous detectors in parallel, one in quadrature with the other.
A comparison of the two detector outputs then determines the
particular binary pair transmitted.
QPSK demodulator 78
Quadrature Amplitude Modulation (QAM)
More general types of multi-symbol signaling schemes may be
generated by letting ai and bi take on multiple values themselves.
The resultant signals are QAM signals. Therefore QAM is a
combined multi-phase/multi-amplitude signaling scheme.
These signals may be interpreted as having multilevel amplitude
modulation, applied independently on each of the two quadrature
carriers.
79
Four level (16-symbol) QAM constellation
The general QAM signal may also be written:
si (t ) = ri cos(ω c t + θ i )
with the amplitude ri and phase angle θi given by the appropriate
combinations of (ai, bi).
A phase-detector-amplitude-level-detector combination could then
also be used to extract the digital information.
In practice, signal shaping must be used to reduce ISI.
An actual modulator would have the input binary pulses shaped
before modulating the carrier.
Alternatively, the bandpass output signals would be passed through
an appropriate bandpass shaping filter before being transmitted.
As the result of shaping, an individual output symbol, nominally
designed to fit into the interval T seconds long, may now span
several T-second intervals.
80
Binary sequence 1 0 1 1 0 1 0
0.8
0.6
0.4
0.2
Amplitude
-0.2
-0.4
-0.6
-0.8
-1
-2 -1 0 1 2 3 4 5 6 7 8 9 10
Time
A series of sinc pulses corresponding to the sequence 1011010

81
A typical signal at time t, in the current slot, may thus be written as:
⎡ n n ⎤
s (t ) = ∑ ⎢a n h(t − ) cos ω c t + bn h(t − ) sin ω c t ⎥
n ⎣ T T ⎦
h(t) represents the impulse response of the shaping filter.
QAM spectrum with pulse shaping (a)

Base band spectrum (b) QAM spectrum 82
We can determine the particular form of QAM, and the type of
shaping required, for transmitting specified bit rates over various
channels.
If the transmission bandwidth is BT hertz, then this corresponds to a
baseband bandwidth of B =BT/2 hertz.
We know previously that
1
BT = 2 B = (1 + r )
Ts
Therefore the symbol rate that may be transmitted over a channel
with baseband bandwidth B hertz is
1 2B
Rs = =
Ts (1 + r )
where the roll-off factor r varies from an ideal value of 0 (for ideal
low-pass filtering) to 1 (for raised-cosine filtering).
83
The symbol rate allowable over the equivalent transmission channel
of bandwidth BT hertz is thus
BT
Rs = symbols/s
(1 + r )
For a QAM signal with M = 2k possible symbols or states, the

allowable bit rate is
kBT
Rb = bits/s
(1 + r )
84
kBT
Rb = bits/s
(1 + r )
Given a transmission bandwidth BT, the desired bit rate and a
particular QAM constellation, the roll-off factor may be found.
Why not go on indefinitely increasing the size of the QAM signal
constellation to achieve indefinitely high bit rate?
As the number of constellation size increases, the phase spacing
between signals reduces correspondingly.
The channel noise and phase jitter makes it more difficult to
distinguish individual points in a constellation as the number of
point increases. This will produce more errors at the receiver.
There is thus a limit on the number of QAM states that may be
used.
85
PSK signal constellations (a) M=4 (b) M=8
Shannon channel capacity theorem:

⎛ S⎞
C = B log 2 ⎜1 + ⎟
⎝ N⎠
C = channel capacity (bits/s) 86
Minimum-Shift-Keying (MSK)
In BPSK modulation, phase discontinuities may occur at the bit
transition point.
This would be smoothed out with appropriate signal shaping but

would clearly result in a varying-amplitude signal.
In communication systems with a non-linearity following the signal
generation point this would result in a broadened signal spectrum
and hence increased signal bandwidth. 87
y (t ) = G [ x(t )]
Transfer characteristic of a nonlinear device
88
X(f )
f
-f2 -f1 0 f1 f2
(a)
Y(f )
f
-2f2 -(f1+f2) -2f1 -f2 -f1 -(f2-f1) 0 f2-f1 f1 f2 2f1 f2+f1 2f2
(b)
(a) Input spectrum. (b) Output spectrum at the

output of a nonlinear channel.
89
A similar problem arises in the generation and transmission of
QPSK (4-phase PSK) signals.
These signals are commonly used in satellite transponders because
of their reduced bandwidth requirements.
Amplifiers in satellite systems are operated near their power
saturation point for better efficiency (e.g., Class C amplifiers).
This is the highly non-linear region and will result in a extraneous
sidebands when passing a signal with amplitude fluctuations (due
to a mechanism called AM-to-PM conversion).
These sidebands deprive the information signals of some of their
portion of transponder power and interfere with nearby channels.
Other high-power communication systems pose a similar problem.
It is thus of interest to consider constant envelope modulation.
In QPSK, the main lobe contains 90% of the signal energy.
90
The sidelobes are smaller than the main lobe by only 14 dB.
The considerable power outside the main lobe will be a source of
serious interchannel interference when QPSK is to be used for
multi-channel communication on adjacent carriers. 91
The large sidelobes are due to the character of the baseband signal.
This signal consists of abrupt changes (e.g., using rectangular
pulse), and abrupt changes give rise to spectral components at high
frequencies.
Recall this:
92
The problem of interchannel interference in QPSK is so serious that
regulatory and standardization agencies will not permit these
system to be used except with appropriate pulse shaping at the
transmitter to suppress the side lobes.
However…the filtering does not necessarily resolve the problem!
In QPSK, the odd and even pulse streams are both transmitted at
the rate of 1/2T bits/s and are synchronously aligned.
Due to this alignment between dI(t) and dQ(t), the carrier phase can
change only once every 2T.
The phase change can be as large as 180°.
In Offset QPSK (OQPSK), also called staggered QPSK (SQPSK),
the timing of the pulse stream dI(t) and dQ(t) is shifted such that the
alignment of the two streams is offset by T.
Since only one component can make a transition at one time, the
phase changes are limited to 0° and ±90° every T seconds.
93
dk(t)
1
d0 d1 d5 d6 d7
t
d2 d3 d4
-1
0 T 2T 3T 4T 5T 6T 7T 8T
dI(t)
1
d0 d6
t
d2 d4
-1
0 2T 4T 6T 8T
dQ(t)
1
d1 d5 d7
t
d3
-1
0 2T 4T 6T 8T
s(t)
d0 = 1 d2 = -1 d4 = -1 d6 = 1
d1 = 1 d3 = -1 d5 = 1 d7 = 1
-1
0 2T 4T 6T 8T
QPSK (constant envelope using 94

rectangular pulses)
dk(t)
1
d0 d1 d5 d6 d7
t
d2 d3 d4
-1
0 T 2T 3T 4T 5T 6T 7T 8T
dI(t)
1
d0 d6
t
d2 d4
-1
0 2T 4T 6T 8T
dQ(t)
1
T
d1 d5 d7
t
d3
-1
0 2T 4T 6T 8T
d2 = -1 d4 = -1 d6 = 1
d0 = 1
d1 = 1 d3 = -1 d5 = 1 d7 = 1
s(t) t
-1
0 2T 4T 6T 8T
OQPSK (constant envelope

using rectangular pulses)
95
For rectangular-shaped data pulses, the envelope of the
QPSK/OQPSK signal is constant.
The rectangular data produce a (sincx)2-type power spectrum that
has large undesirable spectral sidelobes.
96
Now it turns out that when QPSK/OQPSK waveforms with abrupt
phase changes, are filtered so suppress sidebands, the effect of the
filter, at the times of the abrupt phase changes, is to cause
substantial changes in the amplitude of the waveform.
If these signals are used in satellite channels employing highly
nonlinear amplifiers, the constant envelope will tend to be restored
since the nonlinearity suppresses the amplitude variations.
However, at the same time, all of the undesirable frequency side-
lobes, which can interfere with nearby channels and other
communication systems, are also restored.
This suggests that further improvement is possible if the OQPSK
format is modified to avoid phase transitions.
In other words, we need an alternative QPSK scheme which
possesses constant envelop and maintains phase continuity.
Minimum shift keying (MSK) is one such scheme.
97
98
MSK can be viewed as either a special case of continuous-phase
FSK (CPFSK), or a special case of OQPSK with (smoother)
sinusoidal symbol weighting.
99
Mathematically, we can write the MSK signal as
⎛ πt ⎞ ⎛ πt ⎞
si (t ) = ai cos⎜ ⎟ cos ωc t + bi sin ⎜ ⎟ sin ωc t , ai , bi = ±1
⎝T ⎠ ⎝T ⎠
Note that the effect of the sinusoidal weighting terms is to multiply
each carrier by a term going to zero at the bit transition point.
If the carrier has an integer number of half cycles within the symbol
interval T, then there is no phase discontinuity at bit transition points.
This says that we should set
m 1 mR
fc = =
2T 4
with m an integer, and R = 2/T the input bit rate.
One may deduce the following facts from MSK waveforms:
1. The waveform si(t) has constant envelope;
2. There is phase continuity in the RF carrier;
100
101
As we may expect, the sidelobes generated by these smoother
waveforms will be smaller than those associated with the
rectangular waveforms and hence easier to suppress as is required
to avoid interchannel interference.
The MSK baseband signal has a broader first lobe, with the first
zero crossing at f = 0.75R = 3/2T.
The actual MSK signal thus has a first lobe bandwidth, centered
about the carrier frequency fc, of BT = 1.5R = 3/T.
The QPSK signal, has a corresponding baseband first zero crossing
of f =0.5R = 1/T. Its first-lobe transmission bandwidth, centered
about fc is BT = R = 2/T.
The higher frequency content of the MSK signal drops off more
rapidly, however.
The bandwidth of MSK to the 99-percent power point is 1.2R =
2.4/T, while that for the QPSK signal is 8R = 16/T.
102
103

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Baseband Modulation

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Chapter 3 Baseband Pulse Transmission and

Digital Modulation Techniques

 3.2 Transmission Over Band-Pass Channel

 Unipolar Signaling : binary 1 is represented by a high

Time interval over which the Distortion of

 In the presence of ISI and channel noise, traces from the

Eye pattern for binary signalling

Eye patterns for L= 4 signal

Noiseless and ISI-free channel Noisy and ISI channel

Scatter plot for an 8-PSK signal

 These repeaters amplify and “clean up” the signal periodically.

 The amplifying filter increases the amplitude of the low-level input

 Systems are designed so that the sync is derived either

Frequency selective fading channel characteristics

where h(t)= ∏(t/Ts) and an may take on any of the allowed L

 The equivalent system transfer function is

where ⎡ ⎛ t ⎞⎤ ⎛ sin πTs f ⎞

A series of sinc pulses corresponding to the sequence 1011010

 To determine the baud rate that can be supported by the

 The result is that the telephone channel is random in the sense of

 The output yn of the tapped-delay-line equalizer in response to the

 Where wk is the weight at the kth tap. 43

Signal-flow graph of the

 During the training mode, a known sequence is transmitted and a

(a) Spectrum of modulating signal (b) Spectrum of AM wave 53

Example with sinusoidal roll-off shaping

The transmission bandwidth is

Therefore, the transmission bandwidth is

 The single-line spectral plot representing cosωct is thus replaced

(a) periodic ASK signal, (b) Spectrum (positive frequency only)

 A l corresponds to frequency f1, a zero to frequency f2.

 Note, however, that this may also be visualized as the linear

FSK spectrum, sinusoidal roll-off shaping (positive frequency only)

 Note that the FM transmission bandwidth is generally much

 Narrowband FM systems correspond to β << 1.

 The PSK signal has the same double-sideband characteristic as

QPSK signal constellation 76

A series of sinc pulses corresponding to the sequence 1011010

QAM spectrum with pulse shaping (a)

 For a QAM signal with M = 2k possible symbols or states, the

 Shannon channel capacity theorem:

 This would be smoothed out with appropriate signal shaping but

Transfer characteristic of a nonlinear device

(a) Input spectrum. (b) Output spectrum at the

QPSK (constant envelope using 94

OQPSK (constant envelope

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3.2 Transmission Over Band-Pass Channel

Unipolar Signaling : binary 1 is represented by a high

In the presence of ISI and channel noise, traces from the

These repeaters amplify and “clean up” the signal periodically.

The amplifying filter increases the amplitude of the low-level input

Systems are designed so that the sync is derived either

The equivalent system transfer function is

To determine the baud rate that can be supported by the

The result is that the telephone channel is random in the sense of

The output yn of the tapped-delay-line equalizer in response to the

Where wk is the weight at the kth tap. 43

During the training mode, a known sequence is transmitted and a

The single-line spectral plot representing cosωct is thus replaced

A l corresponds to frequency f1, a zero to frequency f2.

Note, however, that this may also be visualized as the linear

Note that the FM transmission bandwidth is generally much

Narrowband FM systems correspond to β << 1.

The PSK signal has the same double-sideband characteristic as

For a QAM signal with M = 2k possible symbols or states, the

Shannon channel capacity theorem:

This would be smoothed out with appropriate signal shaping but