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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

TLC 503
pulses:

Digital Communications x p (t ) 


  
 t  nTs 
n   
Dr Imran Shafi
ai_case@yahoo.com
Note:
 Fourier Transform of impulse train is another impulse train
 Convolution with an impulse train is a shifting operation

Natural Sampling
If we multiply x(t) by a train  Each pulse in xp(t) has width Ts and amplitude 1/Ts
of rectangular pulses xp(t),
 The top of each pulse follows the variation of the signal being
we obtain a gated waveform
sampled
that approximates the ideal
sampled waveform, known  Xs (f) is the replication of X(f) periodically every fs Hz
as natural sampling or  Xs (f) is weighted by Cn  Fourier Series Coeffiecient
gating (see Figure 2.8)  The problem with a natural sampled waveform is that the tops of the
x s (t )  x (t ) x p (t ) sample pulses are not flat
  It is not compatible with a digital system since the amplitude of each
 x (t ) 
n  
c n e j 2  nf s t sample has infinite number of possible values
 Another technique known as flat top sampling is used to alleviate
X s ( f )  [ x ( t ) x p ( t )] this problem

 
n  
c n [ x ( t ) e j 2  nf s t ]

 
n  
cn X [ f  n f s ]

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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
Sample-and-Hold
and Hold (S/H) Flat top sampling (Time Domain)
operation
x '(t )  x(t ) (t )
 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
xs (t )  x '(t ) * p (t )
 

 p (t ) * x(t ) (t )  p(t ) *  x(t )   (t  nTs ) 
 n  

 Taking the Fourier Transform will result to

X s ( f )  [ x s ( t )]
 

 P ( f )   x ( t )   ( t  nTs ) 
 n   
 1 

 P( f )   X ( f ) *  (f  nff s )  Flat top sampling (Frequency Domain)
 T s n   
 Flat top sampling becomes identical to ideal sampling as the
1
 P( f )
Ts

n  
X ( f  nf s ) width of the pulses become shorter

where P(f) is a sinc function

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Recovering the Analog Signal

 Undersampling and Aliasing
 One way of recovering the original signal from sampled signal Xs(f)  If the waveform is undersampled (i.e. fs < 2B) then there will be
is to pass it through a Low Pass Filter (LPF) as shown below spectral overlap in the sampled signal

The signal at the output of the filter will be

different from the original signal spectrum

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

This is the outcome of aliasing!
 Else we run into some problems and signal
This implies that whenever the sampling condition is not met, an
is not fully recovered
irreversible overlap of the spectral replicas is produced

 This could be due to:

1. x(t) containing higher frequency than were expected
2. An error in calculating the sampling rate
Plot of Aliased Cosine (with 90 degrees phase
shift) and Its Reconstruction

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close all
clear all
fs=8000; 1
f0=1;% 1Hz
fst=4;% 4Hz 0.8

t=0:1/fs:2-1/fs; 0.6
x=sin(2*pi*f0*t); % 1 Hz sine wave
0.4
x2=sin(2*pi*(f0+fst)*t); % 5 Hz sine wave
N=length(x); % total length of signal x(1st signal) 0.2
plot(t,x);
0
hold on
plot(t,x2,'g'); -0.2
xs=x(1:fs/fst:N); % samples of 1st signal at interval of
fs/fst=2000 -0.4

tn=t(1:fs/fst:N); -0.6
stem(tn,xs,'r');
pause -0.8

xsn=x2(1:fs/fst:N); -1
stem(tn,xsn,'c'); 0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2

 Solution 1: 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  Aliasing is prevented by forcing the bandwidth of the sampled
 Th anti-aliasing
The ti li i filtfilter iis simply
i l a LPF with
ith cutoff
t ff ffrequency signal
i l tto satisfy
ti f the
th requirement
i t off the
th Sampling
S li ThTheorem
equal to half the sample rate

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Summary Of Sampling

 Solution 2: Over Sampling and Filtering in the Digital
 Ideal Sampling x s (t )  x (t ) x (t )  x (t )   (t  nTs )
Domain n 
(or Impulse Sampling)
 The signal is passed through a low performance (less costly) 
analog low-pass filter to limit the bandwidth.  
n 
x ( nTs ) (t  nTs )
 Sample the resulting signal at a high sampling frequency.
 Natural Sampling
 The digital samples are then processed by a high 
performance digital filter and down sample the resulting
(or Gating)
x s (t )  x (t ) x p (t )  x (t )  cn e j 2 nf s t
signal.
signal 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

Example 1: Practical Sampling Rates

 Consider the analog signal x(t) given by
x(t )  3cos(50 t )  100sin(300 t )  cos(100 t )  Speech
- Telephone quality speech has a bandwidth of 4 kHz
 What is the Nyquist rate for this signal? (actually 300 to 3300Hz)
Example 2: - Most digital telephone systems are sampled at 8000
 Consider the analog signal xa(t) given by samples/sec
 Audio:
xa (t )  3cos
3 2000 t  5sin
5 i 6000 t  cos12000
12000 t - The highest frequency the human ear can hear is
 What is the Nyquist rate for this signal? approximately 15kHz
 What is the discrete time signal obtained after sampling, if - CD quality audio are sampled at rate of 44,000
fs=5000 samples/s. samples/sec
 What is the analog signal x(t) that can be reconstructed from the  Video
sampled values? - The human eye requires samples at a rate of at
least 20 frames/sec to achieve smooth motion

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2.6 Pulse Code Modulation (PCM)

 Pulse Code Modulation refers to a digital baseband signal that is

generated directly from the quantizer output
 Sometimes the term PCM is used interchangeably with quantization

See Figure 2.16 (Page 80)

Advantages of PCM:
 Relatively inexpensive

 Easily multiplexed: PCM waveforms from different

sources can be transmitted over a common digital
channel (TDM)
 Easily regenerated: useful for long-distance
communication, e.g.g telephone
 Better noise performance than analog system

 Signals may be stored and time-scaled efficiently (e.g.,

satellite communication)
 Efficient codes are readily available

Disadvantage:
 Requires wider bandwidth than analog signals

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2.5 Sources of Corruption in the sampled, Signal to Quantization Noise Ratio

quantized and transmitted pulses  The level of quantization noise is dependent on how close any
particular sample is to one of the L levels in the converter
 Sampling and Quantization Effects
 Quantization (Granularity) 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.
 Channel Effects
 For a speech input, this quantization error resembles a noise-
 Channel Noise
like disturbance at the output of a DAC converter
 Intersymbol Interference (ISI)

2.7 Uniform and Nonuniform Quantization

Signal to Quantization Noise Ratio
 A quantizer with equal quantization level is a Uniform Quantizer
 Each sample is approximated within a quantile interval  The mean-squared value (noise variance) of the quantization error is
 Uniform quantizers are optimal when the input distribution is given by:
uniform
 i.e. when all values within the range are
 2   e 2 p(e)de   e 2  1  de  1  e 2 de
q/2 q/2 q/2
equally likely
q / 2 q q
q / 2   q / 2

q/2 2
 1q e q
3

 Most ADC’s are implemented using uniform quantizers 3 q / 2 12

 Error of a uniform quantizer is bounded by  q  e  q
2 2

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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
 The peak power of the analog signal (normalized to 1Ohms )can be
V
expressed as: q  pp
V 2
V p2   L2 q 2  L
P   pp  
1  2   4  where L = 2n is the number of quantization levels for the converter.
  (n is the number of bits).

 Therefore the Signal

g to Q
Quatization Noise Ratio is given
g by:
y
 Si
Since L = 2n, SNR = 22n or in
i ddecibels
ib l
2 2
S N R q  L 2q / 4  3 L 2  S 
q /12 

 10 log (2 2 n )  6 n dB
 N  dB 10

Nonuniform Quantization
 Nonuniform quantizers have unequally spaced levels
 The spacing can be chosen to optimize the Signal-to-Noise Ratio
 Many signals such as speech have a nonuniform distribution
for a particular type of signal
 It is characterized by:  See Figure on next page (Fig. 2.17)
 Variable step size Basic principle is to use more levels at regions with large probability
 Quantizer size depend on signal size
density function (pdf)
 Concentrate quantization levels in areas of largest pdf
 O use fine
Or fi quantization
ti ti (small
( ll step
t size)
i ) for
f weakk signals
i l and
d
coarse quantization (large step size) for strong signals

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Statistics of speech Signal Amplitudes Nonuniform quantization using companding

 Companding is a method of reducing the number of bits required in
ADC while achieving an equivalent dynamic range or SQNR
 In order to improve the resolution of weak signals within a converter,
and hence enhance the SQNR, the weak signals need to be
enlarged, or the quantization step size decreased, but only for the
weak signals
 But strong signals can potentially be reduced without significantly
degrading the SQNR or alternatively increasing quantization step size
 The compression process at the transmitter must be matched with an
equivalent expansion process at the receiver

Figure 2.17: Statistical distribution of single talker speech signal magnitudes

(Page 81)

 The signal below shows the effect of compression, where the  Basically, companding introduces a nonlinearity into the signal
amplitude of one of the signals is compressed
 This maps a nonuniform distribution into something that more
 After compression, input to the quantizer will have a more uniform closely resembles a uniform distribution
distribution after sampling
 A standard ADC with uniform spacing between levels can be used
after the compandor (or compander)
At the receiver, the signal is  The companding operation is inverted at the receiver
expanded by an inverse
operation
 There are in fact two standard logarithm based companding
The process of COMpressing techniques
and exPANDING the signal is
 US standard called µ-law companding
called companding
 European standard called A-law companding
 Companding is a technique
used to reduce the number of bits
required in ADC or DAC while
achieving comparable SQNR

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Input/Output Relationship of Compander Types of Companding

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

log e 1   (| x | / xmax 
y  ymax sgn( x )
log e (1   )

where
 x and y represent the input and output voltages

  is a constant number determined by experiment

 Logarithmic expression Y = log X is the most commonly
used compander
 This reduces the dynamic range of Y
 In the U.S., telephone lines uses companding with  = 255
 Samples 4 kHz speech waveform at 8,000 sample/sec
 Encodes each sample with 8 bits, L = 256 quantizer levels
 Hence data rate R = 64 kbit/sec
  = 0 corresponds to uniform quantization

A-Law Companding Standard (Europe, China, Russia,

Asia, Africa)
 |x|
 A
xmax | x| 1
 ymax sgn( x), 0 
 (1  A) xmax A
y ( x)  
   | x | 
 1  log e  A 
  xmax   1 |x|
 ymax sgn( x),  1
 (1  log e A) A xmax
where
 x and y represent the input and output voltages

 A = 87.6

 A is a constant number determined by experiment

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2.8 Baseband Modulation (PCM waveform types)

 Recall that analog signals can be represented by a sequence of discrete
samples (output of sampler)
 Samples are converted into bits. But bits are just abstract entities that
have no physical definition
 We use pulses to convey a bit of information, e.g.,

 In order to transmit the bits over a physical channel they must be

transformed into a physical waveform
 A line coder or baseband binary transmitter transforms a stream of bits
into a physical waveform suitable for transmission over a channel
 Line coders use the terminology mark for “1” and space to mean “0”
 In baseband systems, binary data can be transmitted using many kinds of
pulses

Goals of Line Coding (qualities to look for)

 A line code is designed to meet one or more of the following goals:
 There are many types of waveforms. Why?  performance criteria!
 Self-synchronization
 Each line code type have merits and demerits
 The ability to recover timing from the signal itself
 The choice of waveform depends on operating characteristics of a
 That is, self-clocking (self-synchronization) - ease of clock lock
system such as:
or signal recovery for symbol synchronization
 Modulation-demodulation requirements
 Long series of ones and zeros could cause a problem
 Bandwidth requirement
 Low probability of bit error
 Synchronization requirement
 Receiver needs to be able to distinguish the waveform associated
 Receiver complexity, etc.,
with a mark from the waveform associated with a space
 BER performance

 relative immunity to noise

 Error detection capability

 enhances low probability of error

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 Spectrum Suitable for the channel

Line Coder
 Spectrum matching of the channel
 The input to the line encoder is
 e.g. presence or absence of DC level
the output of the A/D converter
 In some cases DC components should be avoided or a sequence of values an that
 The transmission bandwidth should be minimized is a function of the data bit
 Power Spectral Density  The output of the line encoder
 Particularly its value at zero is a waveform:



 PSD of code should be negligible at the frequency near zero

Transmission Bandwidth
s (t )  a
n 
n f (t  nTb )
 Should be as small as possible

 Transparency
 The property that any arbitrary symbol or bit pattern can be
transmitted and received, i.e., all possible data sequence should
be faithfully reproducible
where f(t) is the pulse shape and Tb is the bit period (Tb=Ts/n for n
bit quantizer)
 This means that each line code is described by a symbol mapping
function an and pulse shape f(t)
 Details of this operation are set by the type of line code that is
being used

Commonly Used Line Codes

Unipolar NRZ Line Code
 Polar line codes use the antipodal mapping  Unipolar non-return-to-zero (NRZ) line code is defined by
unipolar mapping
  A, w hen X n  1  A, when X n  1
an   a  Where Xn is the nth data bit
  A, w hen X n  0 n
0, when X n  0
 Polar NRZ uses NRZ pulse shape
 In addition, the pulse shape for unipolar NRZ is:
where Tb is the bit period f (t )    t  , NRZ Pulse Shape
 Polar RZ uses RZ pulse shape
 
 Tb 

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Bipolar Line Codes Manchester Line Codes

 Manchester line codes use the antipodal mapping
 With bipolar line codes a space is mapped to zero and
and the following split-phase pulse shape:
a mark is alternately mapped to -A and +A
 A, when X n  1 and last mark   A  Tb   Tb 
 t 4  t 4 
an   A, when X n  1 and last mark   A f (t )     T 
T
0, when X n  0  b   b 
  2   2 
It
is also called pseudoternary signaling or alternate mark inversion
(AMI)
Either RZ or NRZ pulse shape can be used

Summary of Line Codes Comparison of Line Codes

 Self-synchronization
 Manchester codes have built in timing information because they
always have a zero crossing in the center of the pulse
 Polar RZ codes tend to be good because the signal level always
goes to zero for the second half of the pulse
 NRZ signals are not good for self-synchronization

 Error probability
 Polar codes perform better (are more energy efficient) than
Unipolar or Bipolar codes
 Channel characteristics
 We need to find the power spectral density (PSD) of the line
codes to compare the line codes in terms of the channel
characteristics

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Comparisons of Line Codes  First Null Bandwidth

 Unipolar NRZ, polar NRZ, and bipolar all have 1st null bandwidths
 Different pulse shapes are used of Rb = 1/Tb
 to control the spectrum of the transmitted signal (no DC value,  Unipolar RZ has 1st null BW of 2Rb
bandwidth, etc.)  Manchester NRZ also has 1st null BW of 2Rb, although the
 guarantee transitions every symbol interval to assist in symbol timing spectrum becomes very low at 1.6Rb
recovery
1. Power Spectral Density of Line Codes (see Fig. 2.23, Page 90)
 After line coding, the pulses may be filtered or shaped to further
improve there properties such as
 Spectral efficiency

 Immunity to Intersymbol Interference

 Distinction between Line Coding and Pulse Shaping is not easy

2. DC Component and Bandwidth

 DC Components

 Unipolar NRZ, polar NRZ, and unipolar RZ all have DC components

 Bipolar RZ and Manchester NRZ do not have DC components

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