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Outline
4.1 Digital and Analog Modulations Techniques
4.2 Amplitude Modulation
4.3 Angle Modulation
4.4 Frequency vs. Amplitude Modulations
4.5 Advantages of Digital Modulation
4.6 Performance of a modulation scheme
4.7 Equalization, Diversity, and Channel Coding
4.8 Speech Coding
 !




 
 
 Modulation is the °    
information from a message source in a
manner suitable for transmission
 pt generally involves translating a baseband
message signal (called the source) to a
bandpass signal at frequencies that are very
high when compared to the baseband
frequency
 The bandpass signal is called the =

signal and the baseband message signal is
called the =
  signal
 Modulation may be done by varying the
amplitude, phase, or frequency of a high
frequency carrier in accordance with the
amplitude of the message signal
 =
 is the process of extracting the
baseband message from the carrier so that it
may be processed and interpreted by the
intended receiver (also called the sink)
 The ultimate goal of a modulation technique is
to transport the message signal through a
radio channel with the best possible quality
while occupying the least amount of radio
spectrum
 The three basic modulation schemes are
=°  
 ,  

  , and 
 


  and  belong to    

 For digital modulation digital data (0 and 1) is
translated into an analog signal (baseband
signal)
 Digital modulation is required if digital data
has to be transmitted over a medium that only
allows for analog transmission
 Best example for wired networks is the old
analog telephone system: to connect a
computer to this system a == is needed
 The modem then performs the translation of
digital data into analog signals and vice versa
 Digital transmission is used, for example, in
wired local area networks or within a
computer
 pn wireless networks digital transmission cannot
be used directly
 Here, the binary bit-stream has to be translated
into an analog signal Įrst
 The three basic methods for this translation are
=°     (ASK),   
  (FSK), and 
    (PSK)
 Apart from the translation of digital data into
analog signals, wireless transmission requires
an additional modulation, an analog
modulation that shifts the center frequency of
the baseband signal generated by the digital
modulation up to the radio carrier
 For example, digital modulation translates a
1Mbit/s bit-stream into a baseband signal with
a bandwidth of 1 MHz
 There are several reasons why this baseband
signal cannot be directly transmitted in a
wireless system

 
: For the 1 MHz signal we
don͛t need an antenna some hundred
meters high practically, With 1 GHz,
antennas a few centimetres in length
can be used
    = °   : Using
only baseband transmission, FDM could
not be applied and Analog modulation
shifts the baseband signals to different
carrier frequencies
! = 

: Path-loss,
penetration of obstacles, reflection,
scattering, and diffraction depend heavily
on the wavelength of the signal and
Depending on the application, the right
carrier frequency with the desired
characteristics has to be chose
 
  

=

=
 


   

 Amplitude shift keying
 =°     (ASK): is the most
simple 
=
 scheme
 The two binary values, 1 and 0, are
represented by two  
=° 
 ASK illustrated by a figure in the next slide,
this simple scheme only requires low
bandwidth, but is very susceptible to
interference
 Effects like multipath propagation, noise, or
path loss heavily influence the amplitude
 pn a wireless environment, a constant
amplitude cannot be guaranteed, so ASK is
typically not used for wireless radio
transmission
 ASK can also be applied to wireless infrared
transmission, using a directed beam or diffuse
light

Amplitude shift keying (ASK)

 Frequency shift keying
 A modulation scheme often used for wireless
transmission is    "  (FSK)
 The simplest form of FSK, also called binary
FSK (BFSK), assigns one frequency f1 to the
binary 1 and another frequency f2 to the
binary 0
 A very simple way to implement FSK is to
switch between two oscillators, one with the
frequency f1 and the other with f2, depending
on the input
 To avoid sudden changes in phase, special
frequency modulators with continuous phase
modulation (CPM) can be used
 Sudden changes in phase cause high
frequencies, which is an undesired side-effect

Frequency shift keying (FSK)

 Phase shift keying
 
  "  (PSK) uses shifts in the
phase of a signal to represent data
 The figure in the next slide shows a phase shift
of 180°or ʋ as the 0 follows the 1 (vice versa)
 This simple scheme, shifting the phase by
180°each time the value of data changes, is
also called binary PSK (BPSK)
 A simple implementation of a BPSK modulator
could multiply a frequency f with +1 if the
binary data is 1 and with ʹ1 if the binary data
is 0
 To receive the signal correctly, the receiver
must synchronize in frequency and phase with
the transmitter. This can be done using a
Phase Lock Loop (PLL)
 Compared to FSK, PSK is more resistant to
interference, but receiver and transmitter are
also more complex

Phase shift keying (PSK)

 ! =° 

 pn amplitude modulation, the amplitude of a high
frequency carrier signal is varied in accordance to
the instantaneous amplitude of the modulating
message signal
 pf Accos(2ʋh t)is the carrier signal and m (t) is the
modulating message signal, the AM signal can be
represented as
SAM(t)=Ac[1+m(t) cos(2ʋh t)]
 The modulation index k of an AM signal is defined
as the ratio of the peak message signal amplitude
to the peak carrier amplitude
 For a sinusoidal modulating signal m(t)=(Am/Ac)
cos (2ʋh
t) , the modulation index is given by

Where Am is the modulated and Ac carrier signals

 The modulation index is often expressed as a
percentage, and is called percentage modulation
 For instance , if Am = 0.5 Ac and the signal is said
to be 50% modulated the corresponding
sinusoidal modulating and AM signal
represented by figures in the next slide
 A percentage of modulation greater than 100%
will distort the message signal if detected by an
envelope detector
  
=
 


#=
   $ 
 AM signal may be equivalently expressed as
SAM (t) = Re { g(t) exp (j2ʋh t) }
where g (t) is the complex envelope of the AM
signal given by
g(t)=Ac{1+m(t)}
 The spectrum of an AM signal(   h)
can be shown to be

where ɷ (ͻ) is the unit impulse function, and

M(f) is the message signal spectrum
 °=
=


°=°  

The figures show an AM spectrum for a message
signal whose magnitude spectrum is a triangular
function
 The AM spectrum consists of an impulse at
the carrier frequency, and two sidebands
which replicate the message spectrum
 The sidebands above and below the carrier
frequency are called the upper and lower
sidebands, respectively
 The bandwidth of an AM signal is equal to

where h
is the maximum frequency contained

in the modulating message signal
 The total power in an AM signal can be shown
to be

where * represents the average value

 pf the modulating signal is the
above equation may be simplified as

where is the power in the carrier

signal , is the power in the
modulating signal m(t), and k is the
modulation index
 Example:
A zero mean sinusoidal message is applied to a
transmitter that radiates an AM signal with 10
kW power. Compute the carrier power if the
modulation index is 0.6. What percentage of
the total power is in the carrier?
 ! !  

 The two most important classes of angle
modulation being   =
 and
°
 =

 Angle modulation varies a sinusoidal carrier
signal in such a way that the angle of the
carrier is varied according to the amplitude of
the modulating baseband signal
 pn this method, the amplitude of the carrier
wave is kept constant (this is why FM is called
constant envelope)
 There are a number of ways in which the
phase (t) of a carrier signal may be varied in
accordance with the baseband signal
   =
 (FM) is a form of angle
modulation in which the instantaneous
frequency of the carrier signal is varied
linearly with the baseband message signal
m(t), as shown in equation
 where Ac is the amplitude of the carrier, h is
the carrier frequency, and kf is the frequency
deviation constant (measured in units of
Hz/volt)
 pf the modulating signal is a   of
amplitude Am , and frequency h
, then the FM
signal may be expressed as

ȴhh Am
 Where ȴh is the peak frequency deviation of
the transmitter
 For sinusoidal modulating signal with
modulating frequency h

 The frequency =
   ɴf defines
the relationship between the message
amplitude and the bandwidth of the
transmitted signal, and is given by

 Where Am is the peak value of the modulating

signal, ȴh is the peak frequency deviation of
the transmitter and W is the maximum
bandwidth of the modulating signal
 pf the modulating signal is a low pass signal, as
is usually the case, then W is equal to the
highest frequency component hmax =h
present
in the modulating signal, hence
 Example
A sinusoidal modulating signal,
m(t) = 4cos2ʋ4 x 103t, is applied to an FM
modulator that has a frequency deviation
constant gain of 10 kHz/V. Compute
(a) The peak frequency deviation, and
(b) The modulation index
 
 =
 (PM) is a form of angle
modulation in which the angle (t) of the
carrier signal m(t) is varied linearly with the
baseband message signal as shown in
equation

Where kë is the phase deviation constant

(measured in units of radians/volt)
 From the above equations, it is clear that an
FM signal can be regarded as a PM signal in
which the modulating wave is integrated
before modulation
 Yote that an FM signal can be generated by
first integrating m (t) and then using the result
as an input to a phase modulator, conversely a
PM wave can be generated by first
differentiating m(t) and then using the result
as the input to a frequency modulator
 The phase modulation index ɴp , is given by

 Where ȴ is the peak phase deviation of the

transmitter
 ! ! 
  
=° 

 Frequency modulation (FM) is the most
popular

 =
 technique used in
mobile radio systems
 pn FM, the amplitude of the modulated
carrier signal is kept constant while its
frequency is varied by the modulating
message signal
 Thus, FM signals have all their information in
the phase or frequency of the carrier
 This provides a nonlinear and very rapid
improvement in reception quality once a
certain minimum received signal level, called
the FM threshold, is achieved
 pn amplitude modulation (AM) schemes, there
is a linear relationship between the quality of
the received signal and the power of the
received signal since AM signals superimpose
the exact relative amplitudes of the
modulating signal onto the carrier
 Thus, AM signals have all their information in
the amplitude of the carrier
 FM offers many advantages over amplitude
modulation (AM), which makes it a better
choice for many mobile radio applications
 Frequency modulation has Ñ 
==  when compared to amplitude
modulation
 Since signals are represented as frequency
variations rather than amplitude variations,
FM signals are  °Ñ  

=°
 =°  , which tend
to cause rapid fluctuations in the amplitude of
the received radio signal
 Also, message amplitude variations do not
carry information in FM, so burst noise does
not affect FM system performance as much as
AM systems, provided that the FM received
signal is above the FM threshold
 Unlike AM, in an FM system, the modulation
index, and hence bandwidth occupancy, can
be varied to obtain 
 
%% 
°=

 Under certain conditions, the FM signal-to-
noise ratio improves 6 dB for each doubling of
bandwidth occupancy
 This ability of an FM system to trade
bandwidth for SYR is perhaps the most
important reason for its superiority over AM
 However, AM signals are able to occupy less
bandwidth as compared to FM signals, since
the transmission system is linear
 An FM is a constant envelope signal, due to
the fact that the envelope of the carrier does
not change with changes in the modulating
signal
 Hence the transmitted power of an FM signal
is constant regardless of the amplitude of the
message signal
 The issue of amplifier efficiency is extremely
important when designing portable subscriber
terminals since the battery life of the portable
is tied to the power amplifier efficiency
 The capture effect is a direct result of the
rapid nonlinear improvement in received
quality for an increase in received power
 pf two signals in the same frequency band are
available at an FM receiver, the one appearing
at the higher received signal level is accepted
and demodulated, while the weaker one is
rejected
 This inherent ability to pick up the strongest
signal and reject the rest makes FM systems
very 
  %
    and
provides excellent subjective received quality
 pn AM systems, on the other hand, all of the
interferers are received at once and must be
discriminated after the demodulation process
 ! 





 Modern mobile communication systems use
digital modulation techniques
 Advancements in very large-scale integration
(VLSp) and digital signal processing (DSP)
technology have made digital modulation
more   than analog transmission
systems
 Digital modulation offers many advantages
over analog modulation
 Some advantages include 
 
==  and Ñ  to channel
impairments, 
 = °    of various
forms of information (e.g., voice, data, and
video), and 
 
 Digital transmissions accommodate digital
error-control codes which  and/or
 
= , and °°
=°  
   
 ° 
techniques such as source coding, encryption,
and equalization to improve the performance
of the overall communication link
 Yew multipurpose programmable digital
signal processors have made it possible to
implement digital modulators and
demodulators completely in software
 pnstead of having a particular modem design
permanently frozen as hardware, embedded
software implementations now allow
alterations and improvements without having
to redesign or replace the modem
 pn digital wireless communication systems, the
modulating signal (e.g., the message) may be
represented as a time sequence of symbols or
pulses, where each symbol has m finite states
 Each symbol represents n bits of information,
where n = log2m bits/symbol
 ! =

=
 
=
 The performance of a modulation scheme is
often measured in terms of its Ñ  
,
°#   and Ñ
#  
 #   describes the ability of a
modulation technique to preserve the fidelity
of the digital message at # °#  
 pn a digital communication system, in order to
increase noise immunity, it is necessary to
increase the signal power
 However, the amount by which the signal
power should be increased to obtain a certain
level of fidelity(i.e., an acceptable bit error
probability) depends on the particular type of
modulation employed
 The power efficiency, wp , (sometimes called
energy efficiency) of a digital modulation
scheme is a measure of how favourably this
trade-off between fidelity and signal power is
made
 pt is often expressed as the ratio of the signal
energy per bit to noise power spectral density
(Eb/Y0) required at the receiver input for a
certain probability of error (say 10-5)
 r
#   describes the ability of a
modulation scheme to accommodate data
within a = Ñ
#
 pn general, increasing the data rate implies
decreasing the pulse width of a digital symbol,
which increases the bandwidth of the signal
 Bandwidth efficiency reflects how efficiently
the allocated bandwidth is utilized and is
defined as the ratio of the throughput data
rate per Hertz in a given bandwidth
 pf R is the data rate in bits per second, and B is
the bandwidth occupied by the modulated
1W signal, then bandwidth efficiency wB is
expressed as
 The = 
°
 of a digital mobile
communication system is directly related to
the Ñ
#   of the modulation
scheme, since a modulation with a greater
value of wB will transmit more data in a given
spectrum allocation
 There is a fundamental upper bound on
achievable bandwidth efficiency
 Shannon's channel coding theorem states that
for an arbitrarily small °Ñ
Ñ   ,
the maximum possible bandwidth efficiency is
limited by the noise in the channel, and is
given by the channel capacity formula

 where C is the channel capacity (in bps), B is

the RF bandwidth, and S/Y is the signal-to-
noise ratio
 Example
pf the SYR of a wireless communication link is
20dB and the RF bandwidth is 30kHz,
determine the maximum theoretical data rate
that can be transmitted. Compare this rate to
the U.S. Digital Cellular Standard which has a
data rate of 48.6 kbps
 Example
Calculate the RF bandwidth that can be
supported in maximum theoretical data rate
120kbps channel for SYR = 10 dB
 ! &
'
 ((

)
 ) 
 Mobile communication systems require signal
processing techniques that improve the link
performance in hostile mobile radio
environments
 Equalization, diversity, and channel coding are
three techniques which can be used
independently or in tandem to =°
 


 Equalization compensates for inter-symbol
interference (pSp) created by multipath within
time dispersive channels
 pf the modulation bandwidth exceeds the
coherence bandwidth of the radio channel, pSp
occurs and modulation pulses are spread in
time
 An equalizer within a receiver compensates
for the average range of expected channel
amplitude and delay characteristics
 Equalizers must be adaptive since the channel
is generally unknown and time varying
 Diversity is another technique used to
compensate for fading channel impairments,
and is usually implemented by using two or
more receiving antennas
 As with an equalizer, the quality of a mobile
communications link is improved without
increasing the transmitted power or
bandwidth
 While equalization is used to counter the
effects of time dispersion (pSp), diversity
usually employed to reduce the depth and
duration of the fades experienced by a
receiver in a flat fading (narrowband) channel
 One of the best techniques to mitigate the
effects of fading is diversity combining of
independently fading signal paths
 Diversity combining exploits the fact that
independent signal paths have a low
probability of experiencing deep fades
simultaneously
 Thus, the idea behind diversity is to send the
same data over independent fading paths
 These independent paths are combined in
such a way that the fading of the resultant
signal is reduced
 For example, consider a system with two
antennas at either the transmitter or receiver
that experience independent fading
 pf the antennas are spaced sufficiently far
apart, it is unlikely that they both experience
deep fades at the same time
 By selecting the antenna with the strongest
signal, a technique known as selection
combining, we obtain a much better signal
than if we had just one antenna
 Diversity techniques can be employed at both
base station and mobile receivers
 The most common diversity technique is
called spatial diversity, whereby multiple
antennas are strategically spaced and
connected to a common receiving system
 While one antenna sees a signal null, one of
the other antennas may see a signal peak, and
the receiver is able to select the antenna with
the best signal at any time
 Other diversity techniques include antenna
polarization diversity, frequency diversity, and
time diversity
 CDMA systems often use a >& ,
which provides link improvement through
time diversity
 By ͞coding͟ is meant the purposeful
introduction of additional bits in a digital
message stream to allow correction and/or
detection of bits in the message stream that
may have been received in error
 Channel coding improves mobile
communication link performance by adding
redundant data bits in the transmitted
message
 At the baseband portion of the transmitter, a
channel coder maps a digital message
sequence into another specific code sequence
containing a greater number of bits than
originally contained in the message
 The coded message is then modulated for
transmission in the wireless channel
 Channel coding is used by the receiver to
detect or correct some (or all) of the errors
introduced by the channel in a particular
sequence of message bits
 Because decoding is performed after the
demodulation portion of the receiver,  
can be considered to be a ° 
 
 The added coding bits lowers the raw data
transmission rate through the channel
(expands the occupied bandwidth for a
particular message data rate)
 There are two general types of channel codes:
Ñ " 
   

 Channel coding is treated independently from
the type of modulation used
 Although this has changed recently with the
use of trellis coded modulation schemes that
combine coding and modulation to achieve
large coding gains without any bandwidth
expansion
 The three techniques of equalization, are used
to improve radio link performance (to
minimize the instantaneous bit error rate), but
the approach, cost, complexity, and
effectiveness of each technique varies widely
in practical wireless communication systems
 RAKE Receiver
 pn CDMA spread spectrum systems, the chip
rate is typically much greater than the flat
fading bandwidth of the channel
 Whereas conventional modulation techniques
require an equalizer to undo the inter symbol
interference between adjacent symbols,
CDMA spreading codes are designed to
provide very low correlation between
successive chips
 Propagation delay spread in the radio channel
merely provides multiple versions of the
transmitted signal at the receiver.
 pf these multipath components are delayed in
time by more than a chip duration, they
appear like uncorrelated noise at a CDMA
receiver, and equalization is not required
 A RAKE receiver attempts to collect the time-
shifted versions of the original signal by
providing a separate 
 receiver for
each of the multipath signals
 CDMA systems often use a RAKE receiver,
which provides link improvement through
time diversity
The RAKE(M branch or M-finger) receiver, shown in the
Figure, is essentially a diversity receiver designed
specifically for CDMA, where the diversity is provided
by the fact that the multipath components are
practically uncorrelated from one another when their
relative propagation delays exceed a chip period
 A RAKE receiver utilizes multiple correlators to
separately detect the M strongest multipath
components
 The outputs of each correlator are weighted
to provide a better estimate of the
transmitted signal than is provided by a single
component
 Demodulation and bit decisions are then
based on the weighted out puts of the M
correlators
 A weighting network is used to provide a
linear combination of the correlator output for
bit detection
 The outputs of the M correlators are denoted
as Z1, Z2 ,... and ZM, They are weighted by
ɲ1,ɲ2,͙ɲM respectively
 The overall signal Z' is given by

 The weighting coefficients, am, are normalized

to the output signal power of the correlator in
such a way that the coefficients sum to unity,
as show below
 ! °) 
 Speech coders determines the quality of the
recovered speech and the capacity of the
system
 Service providers are continuously met with
the challenge of accommodating more users
within a limited allocated bandwidth
 Low bit-rate speech coding offers a way to
meet this challenge
 The goal of all speech coding systems is to
transmit speech with the highest possible
quality using the least possible channel
capacity
 With low data rate speech coding, digital
modulation schemes offer high spectral
efficiency for voice traffic
 There is a positive correlation between coder
bit-rate efficiency and the algorithmic
complexity, the more complex an algorithm is,
the more its processing delay and cost of
implementation
 Speech coders differ widely in their
approaches to achieving signal compression
 Based on the means by which they achieve
compression, speech coders are broadly
classified into two categories: *
=
) and 
 Waveform coders essentially strive to
reproduce the time waveform of the speech
signal as closely as possible
 They are designed to be source independent
and can hence code equally well a variety of
signals
 Examples of waveform coders include Pulse
Code Modulation ()), Differential Pulse
Code Modulation ()), Adaptive
Differential Pulse Code Modulation ()),
Delta Modulation (), Continuously Variable
Slope Delta Modulation (), and
Adaptive Predictive Coding ())
 Vocoders on the other hand achieve very high
economy in transmission bit rate and are in
general more complex
 They are based on using a priori knowledge
about the signal to be coded(Signal specific)
 )

°

1. The non-uniform probability distribution of
speech amplitude
2. The nonzero autocorrelation between
successive speech samples
3. The non-flat nature of the speech spectra,
the existence of voiced and Unvoiced
segments in speech
4. The quasi-periodicity of voiced speech signals
are the °° that are most often
utilized in coder design
 The most basic property of speech waveforms
that is exploited by all speech coders is that
they are band limited
 Ñ
Ñ      (° h): The non-
uniform probability density function of speech
amplitudes is perhaps the next most exploited
property of speech
 The pdf of a speech signal is characterized by a
very high probability of near-zero amplitudes,
a significant probability of very high
amplitudes, and a monotonically decreasing
function of amplitudes between these
extremes
 The exact distribution depends on the input
bandwidth and recording conditions
 The two-sided exponential (´
  )
function given in the equation below provides
a good approximation to the  %=
h of
telephone quality speech signals

 This pdf shows a distinct peak at zero

 Short-time pdfs of speech segments are also
single-peaked functions and are usually
approximated as a Gaussian distribution
 
   (ACF): Another very
useful property of speech signals is that there
exists much correlation between adjacent
samples of a segment of speech
 This implies that in every sample of speech,
there is a large component that is easily
predicted from the value of the previous
samples with a small random error
 All differential and predictive coding schemes
are based on exploiting this property
 pt gives a quantitative measure of the
closeness or similarity between samples of a
speech signal as a function of their time
separation and mathematically defined as:

 where x (k) represents the kth speech sample

 The autocorrelation function is often
normalized to the variance of the speech
signal and hence is constrained to have values
in the range {-1,1} with C (0) = 1 signals have
an adjacent sample correlation, C(1) , as high
as 0.85 to 0.9
 # °
    (PSD):The
non-flat characteristic of the power spectral
density of speech makes it possible to obtain
significant compression by coding speech in
the frequency domain
 The non-flat nature of the PSD is basically a
frequency domain manifestation of the
nonzero autocorrelation property
 Typical long-term averaged PSD's of speech
show that high frequency components
contribute very little to the total speech
energy
 This indicates that coding speech separately in
different frequency bands can lead to
significant coding gain
 A qualitative measure of the theoretical
maximum coding gain that can be obtained by
exploiting the non-flat characteristics of the
speech spectra is given by the Spectral
Flatness Measure (SFM)
 The SFM is defined as the ratio of the
arithmetic to geometric mean of the samples
of the PSD taken at uniform intervals in
frequency
 Mathematically,

 where, Sk is the kth frequency sample of the

PSD of the speech signal
 Typically, speech signals have a long-term
SFM value of 8 and a short-time SFM value
varying widely between 2 and 500