CC442 Digital Communications

Prof. Dr. Said E. El-Khamy, Life Fellow IEEE

Spring 2021-2022
 Course Contents
Part I: Baseband Digital Modulation:
1- Sampling- A/D – Analog Pulse Modulation PAM, PWM, PPM
2- PCM - PCM Details- Quantization Noise-
3- Mary PCM- NUQ- DPCM & DM
4- ISI and Pulse distortion- Nyquist waveforms- Raised Cosine
waveforms
Part II. Bandpass Digital Modulation
1-Signal Space- distance- Orthog Signals- antipodal signals
2- Bin Mod: PSK, FSK, ASK - PSD
3- Matched Filter- Correlator Receiver - BER
4-QPSK, OQPSK, pi/4 DQPSK, MSK, M-ary – QAM
5-Spectral and Power Efficiencies- Channel Capacity- Comparison of
Diff Dig Modulations’ Performance
Part III. Introduction to channel coding and information
theory:
1- Intro. Info. Theory - Intro. to Source Coding (Huffman Code)
2- Intro. to Channel Coding (Linear Block & Convolution Codes)
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 PART I.
BASEBAND
DIGITAL MODULATION

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 Analog/digital Converters
Sampling and Pulse Modulation
 Basic Model of Communication

Digital Transmitter Channel Receiver Destination

Source

Transmitter

Source Channel Baseband

Encoder Encoder Signaling

Channel

Modulatio Physical Physical Physical Demodulation

n Transmission Medium Reception

Noise
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Receiver
Optimal Channel Source
Filter Decoder Decoder

• Source Encoder translates the out put of the source in an

efficient manner for communication (e.g., compression).

• Channel Encoder transforms the coded source to enable error

detection and correction at the receiver (e.g., add redundancy).

• Baseband signaling encodes digital information in a sequence of

analog pulses.

• Optimal filter makes “best guess” of transmitted analog pulses.

• Channel decoder inverses operation of channel encoder (i.e., error

detection and correction).
• Source decoder inversesCC442
operation of source
Digital Communications , encoder.
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 Sampling
• Sampling is the processes of converting continuous-time analog signal,
xa(t), into a discrete-time signal by taking the “samples” at discrete-time
intervals
– Sampling analog signals makes them discrete in time but still
continuous valued
• Sampling Rate (or sampling frequency fs):
– The rate at which the signal is sampled, expressed as the number of
samples per second (reciprocal of the sampling interval), 1/Ts = fs
• Nyquist Sampling Theorem (or Nyquist Criterion):
– Statement:
“If a signal is sampled at a rate at least, but not exactly equal to
twice the max frequency component of the waveform, then the
waveform can be exactly reconstructed from the samples without any
distortion”
f s  2 f max

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 xs(t) =x(t) P(t/Ts)
For ideal sampling, P(t/Ts) is a periodic train of delta functions, Hence,

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Sampling

– If Rs < 2B, aliasing (overlapping of the spectra) results

– If signal is not strictly bandlimited, then it must be passed
through Low Pass Filter (LPF) before sampling

• Fundamental Rule of Sampling (Nyquist Criterion)

– The value of the sampling frequency fs must be greater than
twice the highest signal frequency fmax of the signal

• Types of sampling
– Ideal Sampling
– Natural Sampling
– Flat-Top Sampling

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 Ideal Sampling
( or Impulse Sampling)
• Is accomplished by the multiplication of the signal x(t) by the uniform train
of impulses (comb function)
• Consider the instantaneous sampling of the analog signal x(t)

◼ Train of impulse functions select sample values at regular intervals


xs (t ) = x(t )   (t − nTs )
n =−

◼ Fourier Series representation:


1 
2

n =−
 (t − nTs ) =
Ts
e
n =−
jns t
, s =
Ts
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Ideal Sampling ( or Impulse Sampling)

 1   jnst
xs (t ) = x(t )    e
• Therefore, by Fourier Series, we have:

 Ts  n =−
◼ Take Fourier Transform (frequency convolution)

  jnst  1 
1
X s ( f ) = X ( f )*    e  = X ( f )*   e jn s t
 
Ts n =−  Ts n =−

1 
s
X s ( f ) = X ( f ) *   ( f − nf s ), f s =
Ts n =− 2
 
1 1 n
Xs( f ) =
Ts

n =−
X ( f − nf s ) =
Ts

n =−
X( f − )
Ts

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Ideal Sampling ( or Impulse Sampling)

◼This shows that the Fourier Transform of the sampled signal is the
Fourier Transform of the original signal at rate of 1/Ts

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• As long as fs> 2fm,no overlap of repeated replicas X(f - n/Ts) will occur in Xs(f)
• Minimum Sampling Condition:

◼ Sampling Theorem: A finite energy function x(t) can be completely

reconstructed from its sampled value x(nTs) with
𝑓𝑠 − 𝐵 > 𝐵 →
𝑓𝑠 > 2 𝐵
  2 f (t − nTs )  
  sin   
  2Ts 
x(t ) =  Ts x(nTs )  
n =−   (t − nTs ) 
 
 

= T
n =−
s x(nTs ) sin c(2 f s (t − nTs ))

provided that => 1 1 CC442 Digital Communications ,

February 2022, Prof. Dr. Said El-Khamy
= 𝑇𝑠 ≤
𝑓𝑠 2𝐵 13
 Graphical Illustration

◼This means that the output is simply the replication

of the original signal at discrete intervals.
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 Practical Sampling
• In practice we cannot perform ideal sampling
– It is practically difficult to create a train of impulses
• Thus a non-ideal approach to sampling must be used
• We can approximate a train of impulses using a train of very thin
rectangular pulses:


 t − nTs 
x p (t ) =    
n =−   

Note:
◼ Fourier Transform of impulse train is another impulse train
◼ Convolution with an impulse train is a shifting operation
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Natural Sampling
If we multiply x(t) by a train of
rectangular pulses xp(t), we
obtain a gated waveform that
approximates the ideal
sampled waveform, known as
natural sampling or gating
(see Figure 2.8)
xs (t ) = x(t ) x p (t )

= x(t ) 
n =−
cn e j 2 nf s t

X s ( f ) = [ x(t ) x p (t )]

= 
n =−
cn [ x(t )e j 2 nf s t ]

= c
n =−
n X [ f − nf s ]

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• Each pulse in xp(t) has width Ts and amplitude 1/Ts
• The top of each pulse follows the variation of the signal
being sampled
• Xs (f) is the replication of X(f) periodically every fs Hz
• Xs (f) is weighted by Cn  Fourier Series Coeffiecient
• The problem with a natural sampled waveform is that
the tops of the sample pulses are not flat
• It is not compatible with a digital system since the
amplitude of each sample has infinite number of
possible values
• Another technique known as flat top sampling is used to
alleviate this problem

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 Flat-Top Sampling
• Here, the pulse is held to a constant height for the
whole sample period
• Flat top sampling is obtained by the convolution of
the signal obtained after ideal sampling with a unity
amplitude rectangular pulse, p(t)
• This technique is used to realize Sample-and-Hold
(S/H) operation
• In S/H, input signal is continuously sampled and
then the value is held for as long as it takes to for
the A/D to acquire its value

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 Flat top sampling (Time Domain)
x '(t ) = x(t ) (t )
xs (t ) = x '(t )* p(t )
 

= p(t )* x(t ) (t ) = p(t )*  x(t )   (t − nTs ) 
 n =− 
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• Taking the Fourier Transform will result to

X s ( f ) = [ xs (t )]
 

= P( f )   x(t )   (t − nTs ) 
 n =− 
 1 

= P( f )   X ( f ) *   ( f − nf s ) 
 Ts n =− 

1
= P( f )
Ts
 X ( f − nf )
n =−
s

where P(f) is a sinc function

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 Flat top sampling (Frequency Domain)

◼Flattop sampling becomes identical to ideal sampling as the width of

the pulses become shorter

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 Recovering the Analog Signal
• One way of recovering the original signal from sampled signal Xs(f) is to
pass it through a Low Pass Filter (LPF) as shown below

◼ If fs > 2B then we recover x(t) exactly

◼ Else we run into some problems and signal
is not fully recovered

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• Undersampling and Aliasing
– If the waveform is undersampled (i.e. fs < 2B) then there will be
spectral overlap in the sampled signal

◼The signal at the output of the filter will be

different from the original signal spectrum

This is the outcome of aliasing!

◼This implies that whenever the sampling condition is not met, an
irreversible overlap of the spectral replicas is produced

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 Summary Of Sampling

• Ideal Sampling xs (t ) = x(t ) x (t ) = x(t )   (t − nTs )
(or Impulse Sampling) n =−

=  x(nT ) (t − nT )
n =−
s s

• Natural Sampling

(or Gating)
xs (t ) = x(t ) x p (t ) = x(t )  cn e j 2 nf s t

n =−

• Flat-Top Sampling
 

xs (t ) = x '(t )* p(t ) =  x(t )   (t − nTs )  * p(t )
 n =− 
• For all sampling techniques
– If fs > 2B then we can recover x(t) exactly
– If fs < 2B) spectral overlapping known as aliasing will occur

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 Practical Sampling Rates

• Speech
- Telephone quality speech has a bandwidth of 4 kHz
(actually 300 to 3300Hz)
- Most digital telephone systems are sampled at 8000
samples/sec
• Audio:
- The highest frequency the human ear can hear is
approximately 15kHz
- CD quality audio are sampled at rate of 44,000
samples/sec
• Video
- The human eye requires samples at a rate of at least 20
frames/sec to achieve smooth motion
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 Analog Pulse Modulation
• Recall that analog signals can be represented by a sequence of discrete samples
(output of sampler)
• Pulse Modulation results when some characteristic of the pulse (amplitude, width
or position) is varied in correspondence with the data signal

• Two Types:
– Pulse Amplitude Modulation (PAM)
• The amplitude of the periodic pulse train is varied in proportion to the
sample values of the analog signal
– Pulse Time Modulation
• Encodes the sample values into the time axis of the digital signal
• Pulse Width Modulation (PWM)
– Constant amplitude, width varied in proportion to the signal
• Pulse Position Modulation (PPM)
– sample values of the analog waveform are used in determining the
position of the pulse signal

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 PAM
Pulse Amplitude Modulation

Volts

time

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 PDM (a.k.a. PWM)
Pulse Duration Modulation (Pulse Width Modulation)

Volts

time

time
min = largest Negative
max = largest Positive CC442 Digital Communications ,
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 PPM
Pulse Position Modulation

Volts

time

time
min = largest Negative
max = largest Positive CC442 Digital Communications ,
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February 2022, Prof. Dr. Said El-Khamy
• PWM (Pulse-Width modulation)
– The samples of the message signal are used to
vary the duration of the individual pulses.
– PDM is wasteful of power

• PPM (Pulse-position modulation)

– The position of a pulse relative to its
unmodulated time of occurrence is varied in
accordance with the message signal.

s(t ) =  g (t − nT
n = −
s − k p m( nTs )) k p m(t ) max  (Ts / 2)

g (t ) = 0, t  (Ts / 2) − k p m(t ) max

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 PAM
Flat-Top Sampling
⚫ The most common technique for sampling voice in PCM systems is
to a sample-and-hold circuit.
⚫ The instantaneous amplitude of the analog (voice) signal is held as a
constant charge on a capacitor for the duration of the sampling
period Ts.
⚫ This technique is useful for holding the sample constant while other
processing is taking place, but it alters the frequency spectrum and
introduces an error, called aperture error, resulting in an inability to
recover exactly the original analog signal.


s (t ) =  m(nT ) h(t − nT )
n = −
s s

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Recovering the original message signal m(t) from PAM
signal

Where the filter bandwidth is W

The filter output is f s M ( f ) H ( f ) . Note that the
Fourier transform of h(t ) is given by
H ( f ) = T sinc( f T ) exp(− j f T )
amplitude distortion delay = T
2
 aparture effect
Let the equalizer response is
1 1 f
= =
H ( f ) T sinc( f T ) sin( f T )
Ideally the original signal m(t ) can be recovered completely.
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Pulse Width and Pulse Position Modulation

⚫ In pulse width modulation

(PWM), the width of each pulse
is made directly proportional to
the amplitude of the information
signal.

⚫ In pulse position modulation,

constant-width pulses are used,
and the position or time of
occurrence of each pulse from
some reference time is made
directly proportional to the
amplitude of the information
signal.

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 Demodulation of PWM and PPM
Since the width of each pulse in the PWM signal is
directly proportional to the amplitude of the
modulating voltage:
⚫ The signal can be differentiated (to PPM in part a),
then the positive pulses are used to start a ramp, and
the negative clock pulses stop and reset the ramp.
⚫ This produces frequency-to-amplitude conversion (or
equivalently, pulse width-to-amplitude conversion).
⚫ The variable-amplitude ramp pulses are then time-
averaged (integrated) to recover the analog signal.

InvestigateCC442
!!!!!Digital Communications ,
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