Adv. Digital Comm.

(EE6502)

Instructor: Dr. JEHAN ZEB

Sir Syed CASE Institute of Technology 1

Agenda

◼ Review of the last lecture

◼ Distortion less Transmission
◼ Bandwidth
◼ Formatting & Sampling
◼ Quantization

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Adding AWGN noise to a signal

◼ The effect on the detection process of a channel

with additive white Gaussian noise (AWGN) is
that the noise affects each transmitted symbol
independently

◼ The term “additive” means that the noise is

simply superimposed to the signal

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Formatting, Sampling
Quantization & Baseband
Modulation

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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
◼ Therefore, we xs (t ) = x(t )    e
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 means that the output is simply the replication of the original
signal at discrete intervals, e.g

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Ideal Sampling ( or Impulse Sampling)
◼ As long as fs> 2fm,no overlap of repeated replicas X(f - n/Ts)
will occur in Xs(f)
◼ Minimum Sampling Condition: f s − f m  f m  f s  2 f m

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

reconstructed from its sampled value x(nTs) with
  2 f (t − nTs )  
  sin  
 
x(t ) =  Ts x(nTs )  
2Ts

n =−   (t − nT s ) 
 
  
= T s x(nTs ) sin c(2 f s (t − nTs ))
1 1
n =− = Ts 
provided that => fs 2 fm

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◼ Ts is called the Nyquist interval: It is the longest time interval that can
be used for sampling a bandlimited signal and still allow
reconstruction of the signal at the receiver without distortion

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Practical Sampling

◼ In practice we cannot perform ideal sampling

❑ It is not practically possible 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:

Note:
◼ Fourier Transform of impulse train is another impulse train
◼ Convolution with an impulse train is a shifting operation

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 If we multiply x(t) by a
train of rectangular
Natural Sampling pulses xp(t), we obtain a
gated waveform that
approximates the ideal
sampled waveform,
known as natural
sampling or gating
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)

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)

◼Flat top sampling becomes identical to ideal sampling under

the limits when the width of the pulses approach to zero

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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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◼ Anti-Aliasing Analog Filter

❑ All physically realizable signals are not completely

bandlimited
❑ If there is a significant amount of energy in frequencies
above half the sampling frequency (fs/2), aliasing will
occur
❑ Aliasing can be prevented by first passing the analog
signal through an anti-aliasing filter (also called a
prefilter) before sampling is performed
❑ The anti-aliasing filter is simply a LPF with cutoff
frequency equal to half the sample rate

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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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Summary Of Sampling 
◼ Ideal Sampling xs (t ) = x(t ) x (t ) = x(t )
n =−

 (t − nTs )
(or Impulse Sampling)

=  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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Signal to Quantization Noise Ratio
◼ The level of quantization noise is dependent on how
close any particular sample is to one of the L levels in
the converter

◼ For a speech input, this quantization error resembles a noise-

like disturbance at the output of a DAC converter

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Uniform Quantization
◼ A quantizer with equal quantization level is a Uniform
Quantizer
◼ Each sample is approximated within a quantile interval
◼ Uniform quantizers are optimal when the input distribution
is uniform
❑ i.e. when all values within the range are equally likely

◼ Most ADC’s are implemented using uniform quantizers

◼ Error of a uniform quantizer is bounded by − q  e  q
2 2

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 Signal to Quantization Noise Ratio

◼ The mean-squared value (noise variance) of the quantization

error is given by:

 2
q/2 1 q/2 q/2
 2 =  e p(e)de =  e
2
q de =
1
 e 2
de
−q / 2 −q / 2   q −q / 2

1 e q 2 q/2
=q =
3

3 −q / 2 12

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◼ The peak power of the analog signal (normalized to 1Ohms )
can be expressed as:

 V pp
2
V p2   L2 q 2 
P= = = 
1  2   4 
 
◼ Therefore the Signal to Quatization Noise Ratio is given by:

L2 q 2 / 4
SNRq = 2 = 3L2
q /12

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◼If q is the step size, then the maximum quantization error that can
occur in the sampled output of an A/D converter is q
V pp
q=
L

where L = 2n is the number of quantization levels for the

converter. (n is the number of bits).

◼ Since L = 2n, SNR = 22n or in decibels

 S 


= 10log (22n ) = 6n dB
 N dB 10

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 Nonuniform Quantization
◼ Nonuniform quantizers have unequally spaced levels
❑ The spacing can be chosen to optimize the Signal-to-Noise
Ratio for a particular type of signal
◼ It is characterized by:
❑ Variable step size

❑ Quantizer size depend on signal size

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◼ Many signals such as speech have a nonuniform distribution
❑ See Figure on next slide
◼Basic principle is to use more levels at regions with large probability
density function (pdf)
❑ use fine quantization (small step size) for weak signals and
coarse quantization (large step size) for strong signals

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Statistics of speech Signal Amplitudes

Statistical distribution of a person speech signal magnitudes

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 Companding

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 - Law Companding Standard (North & South
America, Japan)

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A - Law Companding Standard (Europe,
China, Russia, Asia, Africa)

A=87.6 is used as a standard value

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 law vs A-Law characteristics
With and Without Compression

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 Sources of Corruption in the
quantized signal
◼ Sampling and Quantization Effects
❑ Quantization Noise: Results when quantization
levels are not finely spaced apart enough to
accurately approximate input signal resulting in
truncation or rounding error.
❑ Quantizer Saturation or Overload Noise: Results
when input signal is larger in magnitude than
highest quantization level resulting in clipping of
the signal.
❑ Timing Jitter: Error caused by a shift in the
sampler position. Can be isolated with stable
clock reference.
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 Transmission BW and the output SNR
▪ In PCM we assign distinct group of “n” binary
digits to each of the “L” quantization levels where,

▪The signal m(t) having Bandwidth “B” Hz requires

minimum of “2B” samples per second to represent
Nyquist ▪ Hence, we require a total of 2nB bps
Signalling
Rate ▪ Given a Band-limited channel of “B” Hz,
the max number of signalling elements that can be
transmitted/second error free is 2B
▪Therefore, 1 Hz can transmit max of 2 pieces of
information per sec, Hence, BWPCM = nB Hz

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Example

A television signal (video and audio) has a bandwidth of 4.5

MHz. This signal is sampled, quantized, and binary coded to
obtain a PCM signal.

a) Determine the sampling rate if the signal is to be sampled at

the rate 20% above the Nyquist Rate.

b) If the samples are quantized into 1024 levels, determine the

number of binary pulses required to encode each sample.

c) Determine the binary pulse rate (bits per second) of the

binary-coded signal, and the minimum bandwidth required to
transmit this signal

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 Commonly Used Line Codes
◼ Polar line codes use the antipodal mapping
 + A, when X n = 1
an = 
 − A, when X n = 0
❑ Polar NRZ uses NRZ pulse shape
❑ Polar RZ uses RZ pulse shape

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Unipolar NRZ Line Code
◼ Unipolar non-return-to-zero (NRZ) line code is defined by
unipolar mapping
+ A, when X n = 1
an =  Where Xn is the nth data bit
0, when X n = 0
◼ In addition, the pulse shape for unipolar NRZ is:
 t 
where Tb is the bit period f (t ) =    , NRZ Pulse Shape
 Tb 

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Bipolar Line Codes
◼ With bipolar line codes a space is mapped to zero
and a mark is alternately mapped to -A and +A
+ A, when X n = 1 and last mark → − A

an = − A, when X n = 1 and last mark → + A
0, when X n = 0

◼It is also called Alternate Mark Inversion (AMI)
◼Either RZ or NRZ pulse shape can be used

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Manchester Line Codes
◼ Manchester line codes use the antipodal mapping and
the following split-phase pulse shape:
 Tb   Tb 
t+ 4  t− 4 
f (t ) =   − T 
 Tb   b 
 2   2 

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Summary of Line Codes

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