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CHAPTER 4: DIGITAL

MODULATION AND
TRANSMISSION

Advantages of Digital Communication

Noise immunity- in digital signal, not necessary to


evaluate precise amplitude, frequency of phase.

Multiplexing- Processing and combining digital


signals using a technique called multiplexing. The
transmission rate can be changed to adapt to
different services

Signal regeneration- More susceptible to additive


noise because digital signal does not need
amplification as in analog.

Error correction- Introduce redundancy into the


information in order to correct any error occurs.
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Disadvantages of digital signal

Require more bandwidth

Additional circuit analog signal must be


converted to the digital pulse and converted back
to analog signal

Precise time synchronization

Incompatible with older transmission technology

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Digital Communication Systems

Figure 4.1 Typical digital communication systems

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Info.
Source

Source
Encoder

Channel
Encoder

Digital
Modulator

Channel

Info.
Source

Source
Decoder

Channel
Decoder

Digital
Demodulator

Information source

Digital

Analog

Source Encoder:

The aim of source coding is to take the source data and make it smaller. Each code
must be unique decodable.

Data compression

Convert Analog to Digital

Channel Encoder:

Error detection and possibly correct errors by introducing redundancy to the stream
of bits. The Channel Encoder will add bits to the message bits to be transmitted
systematically. After passing through the channel, the Channel decoder will detect
and correct the errors. A simple example is to send 000 (111 correspondingly)
instead of sending only one 0 (1 correspondingly) to the channel.

Map bit stream to a pulse pattern.

Digital modulator (Wireless)

RF carrier modulation of bits


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Channel

Digital signal can be transmitted through guided


transmission medium such as cable and fiber
optic.

Digital signal can not be transmitted via unguided


transmission medium such free space.. Because
digital signal has infinite bandwidth and can
caused interference with other channel if we
transmit it directly.

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Digital Communication Systems

Figure 4.2 Basic digital communication transformations


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Analog to Digital Converter

Three process involved during the analog-to-digital


conversion:

Sampling

Quantization

Encoding

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Sampling Theorem

The process of changing the continuous time signal


into a discrete signal.

Amplitude values at specific time interval (sampling


rate) are taken using sample-and-hold operation.

Output of sampling process is called Pulse Amplitude


Modulation (PAM) because the successive output
intervals can be described as sequence of pulses with
amplitudes derived from the input waveform samples.

Ts 2Ts 3Ts 4Ts 5Ts 6Ts 7Ts 8Ts 9Ts10Ts11T 12Ts


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For uniform sampling, the sampling rate is constant at fs =


1/Ts

Sampling rate is restricted by the Nyquist criterion,

fs 2fm (Hz),
Ts 1/(2fm) sec,
fm message signal frequency

The sampling rate of fs = 2fm is called the minimum


Nyquist Rate.

Nyquist criterion is the theoretically sufficient condition to


allow an analog signal to be reconstructed completely
from the discrete time samples. Otherwise the signal
becomes indistinguishable (aliasing)
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Figure 4.4 Spectra for various sampling rates. (a) Sampled spectrum
(fs > 2fm). (b) Sampled spectrum (fs < 2fm).

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There are 3 types of sampling techniques:


Impulse

sampling

Natural

sampling

Flat

Top sampling

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Impulse Sampling

Impulse sampling can be performed by multiplying


the signal x(t) with impulse train t nT of period
T. Here, the amplitude of impulse changes with
respect to amplitude of input signal x(t). The output
of sampler is given by

y (t ) x (t ) impulse train
x (t )

(t nT )

y (t ) y (t )

x(nt ) (t nT )

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Natural Sampling

Natural sampling is similar to impulse sampling,


except the impulse train is replaced by pulse train
of period T, i.e. multiply input signal x(t) to pulse
train P t nT as shown below

The output sampler is


y (t ) x(t ) pulse train
x(t ) p (t )
y (t ) x(t )

P(t nT )

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Flat Top Sampling

The top of the samples are flat (constant amplitude).

Use sample and hold circuit for sampling process

Theoretically, the sampled signal can be obtained by


convolution of rectangular pulse p(t) with ideally sampled signal
say y(t) as shown in the diagram

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Example 6.1

Audio signals are often sampled at different rates. CD quality


audio is sampled at 44.1kHz rate while telephone quality audio
sampled at 8kHz. What are the maximum frequencies in the
input signal that can be fully recovered for these two sampling
rates? Briefly describe the theory you use to obtain the results?

Answer:
1.

CD quality audio, the maximum frequency: 44,100Hz / 2 =


22,050Hz.

2.

Telephone quality audio, the maximum frequency: 8kHz / 2 =


4kHz.

3.

This is based on Nyquist theorem: the sampling frequency for


a signal must be at least twice the highest frequency
component in the signal.
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Example 6.2

If an arbitrary (randomly chosen) input signal is


directly sampled, what is the undesired effect may
result and how to solve this?

Answer
1.

This may result in aliasing effect.

2.

To solve this, add an analog low pass filter before


sampling to eliminate high frequency components.

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Example 6.3

What is the Nyquist sampling rate for each of the


following signals?
a)

A low-pass signal with bandwidth of 300 KHz?

b)

A band-pass signal with bandwidth of 200 KHz if the


lowest frequency is 150 KHz?

Answer
a)

In a low-pass signal, the minimum frequency 0. Therefore,


we have fmax = 0 + 300 = 300 KHz. fs = 2 300,000 =
600,000 samples/s

b)

In a bandpass signal, the maximum frequency is equal to the


minimum frequency plus the bandwidth. Therefore, we have
fmax = 150 + 200 = 350 KHz. fs = 2 350,000 = 700,000
samples /s
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Quantization

Quantization is the process of assigning the sampled values into a


finite set of values. The sampled values can only take the specified
values of the quantizing level

The output of the sampler x(nTs) which is discrete in time is fed to


a q-level quantizer. The quantizer compares the input x(nTs) with
it's fixed levels. It assigns any one of the digital level to x(nTs) that
results in minimum distortion or error. The error is called
quantization error, thus the output of the quantizer is a digital
level called q(nTs). The quantized

Quantization
Level

The number of quantization level, L, determined by the number


of bit (n) used by the
system
to encode
eachSYSTEM
sampled signal. (L=2n)
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Resolution

The magnitude difference between adjacent steps.

It is also called quantization interval or quantum

The resolution is equal to the voltage of the


minimum step size, which is equal to the voltage
of the least significant bit.

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Quantization Error (Qe)

Any rounded off errors in the transmitted signal are


reproduced when the code is converted back to analog in
the receiver. This called quantization error Q e.
Qe = |Sampled value Quantized value|

The maximum quantization error is


Qe (max) = resolution / 2

The Qe is equivalent to additive white noise as it alters


the signal amplitude.

Qe is also called quantization noise, Q n


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Figure 6: Natural samples, quantized samples, and PCM


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Example 6.4

Based on Figure 6, determine:


a.

The resolution

b.

The maximum quantization error, Qe(max)

c.

Quantization error for sample number 1,2 and 3.

Answer
a.

Resolution is 1V

b.

The maximum quantization error is,

c.

Qe(max) = resolution/2 = =0.5V

d.

Quantization error for sample number 1, 2, and 3


Sampled value

Quantized value

Quantization error (Qe)

1.3

1.5

0.2

3.6

3.5

0.1

2.5

0.2

2.3

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Signal voltage-to-quantization noise voltage (SQR)

SQR is ratio of signal voltage to quantization noise


voltage.
SQR = V/Qn

The SQR(min) occurs for the lowest magnitude


quantization voltage
SQRmin= Vmin/Qn

The SQR (max) occurs for the maximum magnitude


quantization voltage:
SQRmax=Vmax/Qn
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Dynamic Range (DR)

DR
is the ratio of largest possible quantized voltage to the smallest possible
quantized voltage (other than 0) that can be decoded by digital-to-analog
converter in the receiver.

Vmin = The quantum value (resolution)


Vmax = The maximum quantized voltage that can be discerned by the DAC in the
receiver

In other word, DR is the number of quantization level excluding 0V. Since the
number of bits used for a PCM code depends of the dynamic range. The
relationship between DR and the number of bits in PCM code is

where , n = number of bit used for each sampled signal

DR can be expressed in dB:


DR(dB) = 20 log (DR)

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Example 6.5

For a PCM system with the following parameter


Maximum analog input frequency =4kHz
Maximum decoded voltage at the receiver = 2.55 V
Minimum dynamic range = 46dB
a.

Minimum sample rate

b.

Minimum number of bits used in the PCM code

c.

Resolution

d.

Quantization error

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Answer

a. fs =2fa = 2(4kHz) = 8kHz


b. Determine the absolute value of the minimum dynamic range
46dB = 20log DR
DR = 199.5
2n-1 = 199.5
n = log(199.5+1)/log2 = 7.63
That means 8 bit must be used for the voltage. Since the amplitude range
is2.55kHz, one additional bit, the sign bit is required. Therefore, the
total number of CM bits is nine. Actual DR = 20 log (2n-1) =20log(2561)=48.3dB.
c. Resolution
=Vmax/(2n-1) = 255/(28-1) = 0.01V
d. Quantization error = resolution/2 = 0.005V
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Uniform & Non-uniform Quantization

In uniform quantization (uniform size in quantization level),


quantizing speech signal results in some of the quantization steps to
be wasted.

Quantization noise is the same for all signal magnitudes.

SQR is worse of low level signals than for high level signals

Non-uniform quantization provide fine quantization on low signal


and coarse quantization for high signal.

Achieved by distorting the original signal with a logarithmic compression


characteristic, before applying uniform quantizer

Signal change at small magnitude will carry the uniform quantizer through
more steps than the same change at large magnitude

At the receiver, inverse compression characteristic, called expansion, is


applied so that the overall transmission is not distorted

The processing pair (compression and expansion) is often referred to as


companding. The process are governed by A-Law (Europe) & -Law (US and
Japan) standards.
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Figure 3: Uniform & Non-uniform quantization


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Companding Characteristics

There exist two types of compression


characteristic, - law used in North America and
A-law used in Europe.
y ymax

ln 1 x xmax

-Law
where

ln 1

sgn x

1 x 0
1 x 0

sgn x

A x xmax

1 ln A

A-Law

ymax
y

ymax

sgn x 0

1 ln A x xmax
1 ln A

x
xmax

sgn x

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A
x
1

1
A xmax

Figure 4.11 Compression characteristics. (a) -law characteristic (b) A-law


characteristic
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Encoding

Each quantization level is assigned a specific code word, which


depends on the number of quantization level. For example, For
example, code word length 2 bits (n = 2) will have only 4 levels
whilst code word length 3 bits will have 8 levels, (2n levels).

The output of encoder is the Pulse Code Modulation (PCM) code


word.

For baseband transmission, the codeword bits will then be


transformed into pulse waveform.

The choice of voltage levels is guided by two constraints

The quantize intervals between the levels should be equal

It is convenient for the levels to be symmetrical about zero

Quantization process produce noise, i.e., quantization noise

Increasing number of level reduce quantization noise, but also


increase the number of data needed to be sent.
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Three bit PCM code (Folded binary code)

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Figure 7: Natural samples, quantized samples, and PCM


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PCM encoded signal in binary form:

101 111 110 001 010 100 111 100 011 010 101

Total of 33 bits were used to encode a signal.

Example 6.6

For PCM the bit PCM code (the folded binary code),
determine the quantize voltage, quantization error
(Qe), and PCM code for the analog sample voltage
of +1.07V .

Answer

Quantized level= 1.07 =1


Qe=0.07
PCM code = 101

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Coding Efficiency

Coding efficiency is the numerical indication of


how efficiently a PCM code is utilized.

Coding efficiency is the ratio of minimum number


of bits required to the actual number of PCM bits
used.

coding efficiency

minimum number of bit (including sign bits)


100%
actual number of bits (including sign bits)

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Example 6.7

Based on example 6.5, calculate the coding


efficiency.

Answer
Coding efficiency = 8.63/9x100 = 95.89%

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Figure 5: PCM Transceiver Block Diagram


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Signaling Rate, Bit Rate and Bandwidth in PCM

Sampling rate (SR) = 2 highest signal frequency {fm}


(samples/second) or (Hz)

Bit rate (BR) =SR Number of bits per sample, n (bps)

The BW of binary PCM waveforms depend on the bit rate (BR) and
the waveform pulse shape (sinx/x or rectangular) used to present
the data.

The BW of binary encoded PCM waveform;


BWpcm BR, Unit : Hz

This is obtained only when sinx/x type pulse is used to generate PCM
waveform

However, usually a rectangular type of pulse shape is used so, the Binary
Encoded PCM waveform is:

BWpcm = BR , Unit : Hz
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Example

An analog signal with frequency 40kHz is sampled


at sampling frequency equal to the minimum
Nyquist sampling frequency. Each sample is
encoded using 4-bits PCM. Determine the bit rate.

Answer
Sampling frequency, fs= 2fm = 2x40kHz = 80kHz
BR = 4x80k = 320kbps

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Example 6.8

The bit rate of this Pulse Code Modulation (PCM)


transmission is fixed at 63 kbps. Calculate the
bandwidth

Answer
BW = BR = 1/2x63k = 31.5kHz

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Example 6.9

An analog signal with frequency 4kHz is sampled at


sampling frequency 10% higher than the minimum
Nyquist sampling frequency. Each sample is encoded
with 3-bits PCM code before it transmitted into the
channel. Determine
a.

Bit rate, BR

b.

Bandwidth of the PCM waveforms.

Answer
a.

Determine the sampling rate, Nyquist frequency


fs=2fm = 2(4k) = 8kHz

The actual sampling frequency is 10% higher=fs + 0.1 (fs) =


8k + 0.8k = 8.8kHz.The BR is =3 x 8.8kHz = 26.4kbps
b.

Bandwidth = BR/2 = 26.4k/2 = 13.2kHz


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Other types of pulse modulation

Pulse Width Modulation (PWM)

Pulse Amplitude Modulation (PAM)

Pulse Position Modulation (PPM)

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Pulse Width Modulation (PWM)

The width of a constant amplitude pulse is varied


proportional to the amplitude of the analogue
signal at the time the signal is sampled.

Maximum analogue amplitude gives widest pulse,


and minimum amplitude gives narrowest pulse.

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Pulse Amplitude Modulation (PAM)

The amplitude of a constant width, constantposition pulse is varied according to the amplitude
of the sample of the analog signal.

PAM waveform resemble the original analog signal.

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Pulse Position Modulation (PPM)

The position of the constant-width pulse within a


prescribed time slot is varied according to the
amplitude of the signal at the time the signal is
sampled.

The highest amplitude produces a pulse further to


the far right, lowest amplitude produces a pulse
further to the far left.

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Figure 4.12
Pulse modulation:
(a) analogue signal;
(b) sample pulse;
(c) PWM;
(d) PPM;
(e) PAM;
(f) PCM

Baseband Transmission

PCM transform analogue signals into binary digits, has to be translate


into electrical pulse before transmission in baseband channel.

The simplest representation is assigning a pulse to bit 1 and absence of


a pulse to bit 0 (unipolar).

Each pulse carry energy (area under the pulse) necessary for detection,
advantageous to have pulse width, T, equal to bit time T.

Another way is to assign two polarity (bipolar), i.e., positive polarity for
binary 1 and negative polarity for binary 0. Thus, described as
sequence of changing polarity.

Figure 4.13 shows basic waveform representation of binary signal.

Assigning the pulse depending on the previous state of pulse or change


in binary symbol state also possible, termed differential encoding.

Figure 4.14 shows other possible waveform representation of PCM code.


Can be divided into four groups,
1. Nonreturn-to-zero (NRZ)
3. Phase encoded

2. Return-to-zero (RZ)

4. Multilevel binary
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Figure 4.13 Waveform


representation of binary digits.
(a) PCM sequence.
(b) Pulse representation of PCM.
(c) Pulse waveform (transition
between two levels)

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PCM waveforms

Figure 4.14 Various PCM waveforms


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PCM waveforms

Non Return To Zero (NRZ) Waveform

NRZ L (L for Level)

Binary 1 is represented by one voltage level and binary 0


by another voltage level.

NRZ M (M for Mark or 1)

Mark or 1 is represented by change in voltage level.

Space or 0 is represented by no change in voltage level.

NRZ S (S for Space or 0)

Compliment of NRZ M.

Space is represented by change in voltage level,

Mark is represented by no change in voltage level.

Both NRZ M and NRZ S referred to as differential


encoding.
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PCM waveforms (cont.)

Return to Zero (RZ) waveform

Unipolar RZ

1 half bit wide pulse

0 absence of pulse

Bipolar RZ

1 and 0 represented by opposite-level pulse that are onehalf bit wide.

Pulse in each bit interval.

RZ AMI (Alternate Mark Inversion)

1 is represented by equal amplitude alternating pulses.

0 is represented by absence of pulses.

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PCM waveforms (cont.)

Phase Encoded

Manchester coding (bi-phase-level)

1 half bit wide pulse position during first half of bit interval.

0 half bit wide pulse position during second half of bit interval.

Bi-phase-Mark

Transition occurs at the beginning of every bit interval.

1 second transition one-half-bit later

0 no second transition

Bi-phase-S

Transition occurs at the beginning of every bit interval.

1 no second transition

0 second transition one-half-bit later

Delay Modulation

1 transition at the middle of the bit interval

0 no transition, unless followed by another 0, at the end of bit


interval.
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PCM waveforms (cont.)

Multi Level Binary Waveform

Uses 3 levels, instead of 2 levels, to encode the binary


data.

Examples are RZ-AMI, Bipolar RZ

Dicode NRZ

Transition from 1 to 0 or 0 to 1 changes the pulse


polarity.

No transition sent zero level.

Dicode RZ

1 to 0 or 0 to 1 transition produces a half-duration


polarity changes,

Otherwise, zero level is sent.


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Why do we need so many PCM waveform?

DC component Eliminating DC energy from the signals power


spectrum enables system to be AC coupled.

Self-Clocking Allows symbol or bit synchronization, aid


recovery of the clock signal.

Error detection some schemes provide means of detecting


errors without additional error-detection bits in data sequence
(duobinary)

Bandwidth compression multilevel codes increase the


efficiency of bandwidth utilization by allowing reduction in
required bandwidth for given data rate; thus more information
per unit bandwidth.

Differential coding allows the polarity of differentially


encoded waveforms to be inverted without affecting the data
detection.

Noise immunity Some of the schemes are more immune to


noise than others.
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Figure 4.15 Spectral densities of various PCM waveforms.

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DIGITAL MODULATION

So far we have covered some basic techniques of


producing digital signal from analog signal.

We have covered baseband signaling, i.e. transmitting


the digital signal at its original bandwidth.

Transmitting signal at its baseband level requires each


transmitter to have unique connection with its receiver.
This can be achieved using dedicated link such as cables.

However, if the signal is to be transmitted through free


space where each system is assigned a specific frequency
spectrum, the baseband signal needs to be translated
into digital radio at the specific frequency spectrum.

This is done via either ASK, FSK, PSK or combination of


the three basic digital modulation techniques.
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Figure 4.17 Basic block diagram for digital modulation.


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Terminology

M-ary Encoding

M simply represents a digit that corresponds to the number of


conditions, levels, or combinations possible for a given
number of binary variables.

Encoding at higher level than binary, result in more


information (bits) sent per unit symbol.

Baud and Minimum Bandwidth

Baud is symbol rate, number of symbols per second.

One symbol or signaling element may represent 1 bit at one


time (0 or 1) or it can be used to represent more than 1
bit.

Bit per second (bps) is higher than baud for M-ary system
because each symbol carries more bit.

In binary system, baud and bps are equal.


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Energy bit over Noise density

Key parameter in digital systems is given by Eb/N0


ratio (Eb = the average bit energy, N0 = the thermal
noise power density). The formulas are:
Eb = PRTb

N0

= kNT
where:
PR = average power in the received signal (Unit: W)
Tb = bit duration (Unit: s)
k = 1.38 10-23 J/K (Boltzmann's constant)
NT = noise temperature referred to the receiver
input (Unit: K)
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Channel Capacity

Nyquist
channel capacity is the theoretical limit of the

maximum capacity that can be achieved:


bps

Minimum bandwidth necessary to pass M-ary digitally


modulated carrier (minimum Nyquist bandwidth in Hz)

B
BW

log
M

let k = log2M, then


where k is the number of bits encoded into each
signaling element.
Baud is the term for symbol rate:
(baud or symbol/s)
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AMPLITUDE SHIFT KEYING (ASK)

Simplest
modulation technique where binary information

signal directly modulates the amplitude of an analog carrier.

Similar to AM except only two output amplitude is possible.


In AM amplitude refers to the envelop of the modulating
signal.
vc (t ) Vc cos(0t )
Let the carrier signal,

Amplitude shift keying then,

is the normalized binary waveform, where +1 for logic 1


and -1 for logic 0. Therefore, for logic 1
and for logic 0
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Thus,
the modulated
wave is either or 0.
Hence, it is sometime
referred as on-off keying
(OOK).

Figure 4.17 shows ASK


signal. Notice the baud
rate and the bps is the
same.
Figure 4.17 ASK signal.

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OOK can be done using a multiplier circuit.

Figure 4.18 OOK multiplier circuit.


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An efficient method to demodulate OOK is to use


synchronous detection (coherent detection):

Figure 4.19 OOK synchronous detection.


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Demodulation of OOK wave can also be done using


an envelope detector (noncoherent or
nonsynchronous detection).

The coherent detector is more complicated than


the envelope detector, but it results in a lower
probability of error for a given signal-to-noise
input.

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error probability for optimum coherent


Bit
detection is:

Eb
1
Pbe erfc
2
2N0

erfc is the short term for complementary error


function:
where erf is the short term for error function.

Tabulation of the error function:


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Example

Given Eb/N0 = 0.32

Eb
1
Pbe erfc
2
2N0

1
1 Eb
1
1
erfc 0.32
Pbe erfc
2
2 N0
2
2
1
1
Pbe erfc 0.16 erfc 0.4
2
2
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erfc 0.4 1 erf 0.4

erfc 0.4 1 0.42839


erfc 0.4 0.57161

1
Pbe erfc 0.4
2
1
Pbe 0.57161
2
Pbe 0.2858

From table

The optimum noncoherent detector requires that


Eb/N0 > 1, & for this condition.

Pbe

e
2

Given Eb/N0 = 1.11 P


be

Pbe
Pbe

Eb
2 N0

Eb
2 N0

Eb
N0

1
1

e
e
2
2
1 12 1.11
1 0.555
e
e
2
2
1
0.5741 0.2870
2

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FREQUENCY SHIFT KEYING (FSK)

Also

called BFSK, since its involves only two frequency (binary).

Similar to FM except only two level of modulating signal.

f peak shift in carrier frequency, which is proportional to the


amplitude of the binary input signal, and the direction of shift is
determine by the polarity.

Therefore, for
for

Bit rate and baud rate of FSK is equal. But the bandwidth can be
approximated by

BW = 2(f +B)
which is similar to Carsons rule except,

fm is replace with B.

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To prevent
discontinuities, the
modulation can be
achieved by frequency
modulating a common
carrier.

Figure 4.20
Generation circuit:
a)
b)

FSK
CFSK

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Figure 4.21 FSK modulator, tb, time of one bit = 1/fb; fm, mark frequency; fs, space
frequency; T1, period of shortest cycle; 1/ T1, fundamental frequency of binary square
wave; fb, input bit rate (bps)
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(a)

(b)
Figure 4.23 (a) Coherent and (b) noncoherent detection for FSK.
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The
probability of bit error for coherent FSK
detector is given by:

Eb
1
Pbe erfc
2
2N0

The probability of bit error for non-coherent FSK


detector is given by:

1
Pbe e
2

Eb
2 N0

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PHASE SHIFT KEYING (PSK)

Angle modulated, constant amplitude digital


modulation.

Similar to phase modulation, but the input is


binary signal and there are limited number of
output phase possible.

Simplest form is BPSK, k = 1, M = 2.

As the digital signal input changes state, the phase


of the output carrier shifts between two angle that
are separated by 180o.

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UP = unipolar
BP = bipolar

Figure 4.24 Generation of BPSK signal.


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Figure 4.25 Recovery of BPSK signal.


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For a BPSK input signal of (logic 1), the output of


the balanced modulator

where
leaving

filtered out

For a BPSK input signal of (logic 0), the output of


the balanced modulator

where
leaving

filtered out
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Figure 4.26 BPSK modulator: (a) truth table; (b) phasor diagram; (c)
constellation diagram
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Figure 4.27 Output phase-versus-time relationship for a BPSK modulator


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The probability of bit error for coherent PSK


detector is given by:

Eb
1
Pbe erfc
2
N0

The probability of bit error for non-coherent PSK


detector is given by:
Eb

1 N0
Pbe e
2

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DIFFERENTIAL PHASE SHIFT KEYING (DPSK)

Phase shifted relative to previous transmission


rather than some reference signal

S0(t) = Spr(t) /2,

S1(t) = Spr(t) 3/2,

where: Spr(t) is the previous bit signal element

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PSK

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DPSK

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QUADRATURE PHASE SHIFT KEING (QPSK)

M-ary encoding scheme with 2 input bit, k=2 and M


= 22 = 4 (output phases). One output phase change
for every 2 input bit.

Produce 4 output phases (-135o,-45o,+135o,+45o,)


with 4 different input bit combination (00, 01,10,
11).

In Figure 5.15 (QPSK modulator), 2 serial input bit


is parallel outputted to the I channel and Q
channel respectively. A QPSK modulator is 2 BPSK
modulator combined in parallel.

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The
I bit modulates a carrier that is in phase with
reference oscillator (I for in phase channel). The Q
bit modulates a carrier that is 900 out of phase with the
reference carrier (Q for quadrature channel).

For a logic 1=+1V and a logic 0=-1 V, 2 phases are possible


at the output of I balance modulator ( and ), and 2
phases at the output of Q balance modulator ( and ).

The 4 possible resultant phasors are:

).

Each phasor has the same amplitude (Different from QAM).


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QPSK modulator

Figure 4.31 QPSK modulator


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Figure 4.32 QPSK modulator: (a) truth table; (b) phasor


diagram; (c) constellation diagram
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Figure 4.33 Output phase-versus-time relationship for a QPSK modulator

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QPSK Receiver

In the QPSK receiver, the power splitter directs the


input QPSK signal to the I and Q product
detectors & carrier recovery circuit.

Carrier recovery circuit reproduces the original


carrier oscillator signal. The recovered carriers
frequency and phase must be coherent with
transmit reference carrier.

QPSK signal is demodulated in the I and Q


product detectors which generates the original I
and Q data bits. The output of product detectors
are fed to combining circuit for conversion from
parallel I and Q to a single binary output data
stream.
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QPSK receiver

Figure 4.34 QPSK receiver

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The
receive QPSK signal, () is one of the inputs to the I
product detector.

The other input is the recovered carrier ().

The output of the I product detector is

(logic 0)
filtered out

equal 0

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The
received QPSK signal, () is one of the input to

the Q product detector.

The other input is the recovered carrier shifted 90 o


in phase ().

The output of the Q product detector is

(logic 0)
filtered out

Equal 0

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