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# PULSE CODE MODULATION & SOURCE

CODING

Sampling Theory

1
Sampling Theory
Signal Reconstruction
Aliasing

LEARNING OBJECTS
2
Basic elements of a PCM system
3
Sampling Theory
In many applications, e.g. PCM, it is useful to represent
a signal in terms of sample values taken at
appropriately spaced intervals.
The signal can be reconstructed from the sampled
waveform by passing it through an ideal low pass filter.
In order to ensure a faithful reconstruction, the original
signal must be sampled at an appropriate rate as
described in the Nyquists sampling theorem.
A real-valued band-limited signal having no spectral
components above a frequency of B Hz is determined
uniquely by its values at uniform intervals spaced no greater
than seconds apart.

4
Sampling Theory
)] ( [ * ) (
2
1
) ( t F G G
s
T s
o e
t
e =
5
Impulse Sampling
6
Impulse Sampling
7
Sampling Visualized in
Frequency Domain
8
Interpolation
From the spectrum of the sampled signal, we can see that the original
signal can be recovered by passing its samples through a LPF
9
Ideal Interpolation
10
Ideal Interpolation
11
Ideal Interpolation
12
Practical Considerations in Nyquist Sampling
13
14
15
Aliasing
Resultantly, they will be not band
limited.
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Aliasing
17
A Solution: The Antialiasing Filter
The anti-aliasing, being an ideal filter, is unrealizable. In
practice we use a steep cutoff which leaves a sharply
attenuated residual spectrum beyond the folding
frequencies.
18
Practical Sampling
19
Practical Sampling
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Some Applications of Sampling Theorem
Sampling theorem is very important in signal analysis,
processing and transmission because it allows to
replace a continuous time signal by a discrete
sequence of numbers. This leads into the area of
digital filtering.
In communication, the transmission of continuous-
time message reduces to the transmission of a
sequence of numbers. This opens the doors to many
new techniques of communicating continuous-time
signals by pulse trains.
The continuous-time signal g(t) is sampled, and
sampled values are used to modify certain parameters
of a periodic pulse train.
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The sampled value can be used to vary amplitude, width or
position of the pulse in proportion to the sample values of the
signal g(t). Accordingly we get
Sampling
g(t)
Pulse
Modulation
Value of
the sample
Some Applications of Sampling
Theorem
[22]
Pulse Modulated Signals
23
Some Applications of Sampling Theorem
Pulse modulation permits simultaneous transmission of several
signals on a time-sharing basis: Time Division Multiplexing.
Because a pulse modulated signal occupies only a part of the
channel time, therefore several pulse modulated signals can be
transmitted on the same channel by interweaving.

Similarly several baseband signals can be transmitted
simultaneously by frequency division multiplexing where
spectrum of each message is shifted to a specific band not
occupied by any other signal.
24
Time Division Multiplexing
25
Pulse Code Modulation
Most useful and widely used of all the pulse modulations.
PCM is a method of converting an analog signal into a digital signal
(A/D conversion).
An analog signals amplitude can take on any value over a continuous
range while digital signal amplitude can take on only a finite number
of values.
An analog signal can be converted into a digital signal by means of
three steps:
sampling
quantizing, that is, rounding off its value to one of the
closest permissible numbers (or quantized levels)
Binary coding, that is conversion of quantized samples to
0s and 1s.
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Amplitude Quantization
{ }
function. staircase a is which
stic, characteri quantizer the called is ) g( mapping The
size. step the is , levels tion reconstruc or tion representa the are
L , 1,2, , where is output quantizer then the ) ( If
) ( amplitude discrete a into ) ( amplitude sample
the ing transform of process The : on quantizati Amplitude
hreshold. decision t or the level decision the is Where
, , 2 , 1 , :
cell partition Define
1

1
m
m m
t m
nT nT m
m
L k m m m
k k
s s
k
k k
=

= e
= s <
+
+
v
v

k J
J
k k
k
k
27

(b) Mid-rise

Scalar Quantizer

28
Quantization
29
Quantization
30
Quantization Error
31
Quantization Error
32
Quantization Noise
33
Quantization Noise
34
Quantization Noise

12

1
) ( ] [

otherwise
2 2

, 0
,
1
) (
levels of number total : ,

2
is size - step the
type midrise the of quantizer uniform a Assuming
) 0 ] [ ( ,

value sample of variable
random by the denoted be error on quantizati Let the
2
2
2
2
2
2
2 2 2
max max
max
A
=
A
= = =

A
s <
A

A
=
<
= A
= =
=
} }
A
A

A
A

<
dq q dq q f q Q E
q
q f
L m m m
L
m
M E V M Q
m q
q Q
Q Q
Q
o
v
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Quantization SNR
). (bandwidth n increasing lly with exponentia increases (SNR)
)2
3
(
) (
) ( of power average the denote Let
2
3
1

2
2

log n
sample per bits of number the is n where
2
form, binary in expressed is sample quatized When the
o
2n
2
max
2
o
2 2
max
2
2
max
m
P
P
SNR
t m P
m
m
L
L
Q
n
Q
R
n
=
=
=
= A
=
=

o
o
, 6dB per bit
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Non-uniform Quantization
Motivation
Speech signals have the
characteristic that small-
amplitude samples occur more
frequently than large-
amplitude ones
Human auditory system
exhibits a logarithmic
sensitivity
More sensitive at small-
amplitude range (e.g., 0
might sound different from
0.1)
Less sensitive at large-
amplitude range (e.g., 0.7
might not sound different
much from 0.8)

histogram of typical
speech signals
[37]
Non-uniform Quantization
38
Non-uniform Quantization
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Non-uniform Quantization
Non-uniform Quantization = Compression + Uniform quantization
40
Non-uniform Quantization
41
Law / A Law
The -law algorithm (-law) is a companding algorithm, primarily
used in the digital telecommunication systems of North America
and Japan.
Its purpose is to reduce the dynamic range of an audio signal.
In the analog domain, this can increase the SNR achieved during
transmission, and in the digital domain, it can reduce the
quantization error (hence increasing signal to quantization noise
ratio).
A-law algorithm used in the rest of worlds.
A-law algorithm provides a slightly larger dynamic range than the -
law at the cost of worse proportional distortion for small signals.
By convention, A-law is used for an international connection if at
least one country uses it.
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Law Compression
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A-Law Compression
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Binary Coding
From practical viewpoint, a binary digital signal
(a signal that can take on only two values) is very
desirable because of its simplicity, economy, and
ease of engineering. We can convert an L-ary
signal into a binary signal by using pulse coding.
This code, formed by binary representation of
the 16 decimal digits from 0 to 15, is known as
the natural binary code (NBC).
Each of the 16 levels to be transmitted is
assigned one binary code of four digits. The
analog signal m(t) is now converted to a (binary)
digital signal. A binary digit is called a bit for
convenience.
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Binary Coding
Now each sample is encoded by four bits.
To transmit this binary data, we need to assign
a distinct pulse shape to each of the two bits.
One possible way is to assign a negative pulse
to a binary 0 and a positive pulse to a binary 1
so that each sample is now transmitted by a
group of four binary pulses (pulse code). The
resulting signal is a binary signal.
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47
Pulse Code Modulation Examples
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Transmission Bandwidth and SNR
For a binary PCM, we assign a distinct group of n binary
digits (bits) to each of the L quantization levels. Because a
sequence of n binary digits can be arranged in distinct 2
n

patterns,
L=2
n
or n=log
2
L

Each quantized sample is, thus, encoded into n bits.
Because a signal m (t) band-limited to B Hz requires a
minimum of 2B samples per second, we require a total of
2nB bits per second (bps), that is, 2nB pieces of
information per second.
Because a unit bandwidth (1 Hz) can transmit a
maximum of two pieces of information per second, we
require a minimum channel of bandwidth Hz, given by
B
T
=nB Hz

This is the theoretical minimum transmission bandwidth
required to transmit the PCM signal.
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Transmission Bandwidth and SNR
We know that L
2
= 2
2n
, and the output SNR can be expressed
as

where

Lathi book
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Transmission Bandwidth and SNR
We observe that the SNR increases exponentially with the
transmission bandwidth BT. This trade of SNR with bandwidth
is attractive. A small increase in bandwidth yields a large
benefit in terms of SNR. This relationship is clearly seen by
rewriting using the decibel scale as
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Transmission Bandwidth and SNR
This shows that increasing n by 1 (increasing one
bit in the code word) quadruples the output SNR
(6-dB increase).
Thus, if we increase n from 8 to 9, the SNR
increases only from 32 to 36 kHz (an increase of
only 12.5%).
This shows that in PCM, SNR can be controlled by
transmission bandwidth.
Frequency and phase modulation also do this. But
it requires a doubling of the bandwidth to
quadruple the SNR. In this respect, PCM is
strikingly superior to FM or PM.
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Differential PCM

53
Differential Pulse Code Modulation (DPCM)
If [ ] is the th sample, instead of transmitting [ ],
difference [ ] [ ] [ 1] is transmitted.
At the receiver, knowing of the difference [ ] and the
previous sample value [ 1], we can construc
m k k m k
d k m k m k
d k
m k
=
t [ ].
Difference between successive samples is generaly much
smaller than the sample values.
m k
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Differential Pulse Code Modulation (DPCM)
2
Therefore the peak amplitude of the transmitted
value reduces considerably. Hence quantization interval
for a given (or ) by .
12

p
p
m
v m
v L n
L
A
A =
For a given (transmission bandwidth), we can
increase the SNR, or for a given SNR, we can reduce
(transmission bandwidth).
n
n
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DPCM
This scheme by estimating
(predicting) the value of th sample [ ] from the
knowledge of the previous sample val
can
u
further e
.
b
es
improved
k m k
At the receiver also we determine the estimate m[k],
from the previous sample values and generate [ ],
m k
d k m k
] [

k m
] [

] [ ] [ k m k m k d =
If the estimate is , then the difference
is transmitted.
] [

k m
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] [

k m
DPCM
Since difference between the predicted value and
the actual value will be even smaller than
the difference between the actual values, this scheme
is kn Differential Pulse Code Modulation own as (DPCM).
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How Does the Predictor Works ?
2 3
.. ...
Suppose for a signal, we can express the as
(
Taylor
) ( ) ( ) ( ) ( ) .............
2! 3!
( ) ( ) for
Serie
s all
s
m
S S
S s
s S
T T
m t T m t T m t m t m t
m t T m t T
-
-
+ = + + + +
~ +
.
If we know the ( ), we can predict the future signal
value from knowledge of signal and its derivative.
Let us denote the th sample of ( ) by [ ], that is
[ ] [ ], and ( ) [ 1] and so o
S S S
m t
k m t m k
m kT m k m kT T m k = =
S
n
setting t=kT
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The Predictor
.
[ ] [ ( ) ( )]/ , then we obtain
[ 1] ( ) [ ( ) [ 1) / ]
2 [ ] [ 1]
S S S S S
S S
m kT m kT m kT T T
m k m k T m k m k T
m k m k
~
+ ~ +
~
Crude prediction of [ 1] can be made by obtaining
two previous samples. This approximation can be
further imrpoved as we add more stages in the series.
m k +
.
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The Linear Predictor
1 2 3
1 2 3
In general, we can express the prediction formula
[ ] [ 1] [ 2] [ 3] ... [ ]
and the predicted value of [ ] is
[ ] [ 1] [ 2] [ 3] .... [ ]
Therefore larger N would resu
N
N
m k a m k a m k a m k a m k N
m k
m k a m k a m k a m k a m k N
~ + + +
~ + + +
lt in better prediction value.
] [

k m
A tapped delay-line (transversal)
filter used as a linear predictor
with tap gains equal to prediction
coefficients
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Linear Prediction Coding (LPC)

Consider a finite-duration impulse response (FIR)
discrete-time filter which consists of three blocks :
1. Set of p ( p: prediction order) unit-delay elements (z
-1
)
2. Set of multipliers with coefficients w
1
,w
2
,w
p
3. Set of adders ( )

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DPCM

The DPCM
transmitter
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SNR Improvement
2
Peak value of ( ) be and ( ) be ( ),
For same value of , quantization step v in DPCM
is reduced by the factor .
Becuase the quantization noise power is ( ) / 12,
the quatization noi
p p
p
p
m t m d t d difference
L
d
m
v
A
A
2
se in DPCM reduces by the factor
( ) , and the SNR increases by the same factor.
p
p
m
d
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By exploiting redundancies from the speech signal, prediction can
be improved
Predictor coefficients are derived from the sampled signal and
transmitted along with the signal
Prediction can be so good that after some time only the predictor
coefficients are sent.
We get transmission at 8-16 kbps with the same quality of PCM
Coded Excited Linear Prediction (CELP)
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Delta Modulation

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Delta Modulation (DM)
Sample correlation used in DPCM is further
exploited in delta modulation (DM) by
over sampling (typically 4 times the Nyquist rate)
the baseband signal.
This increases the correlation between adjacent
samples, which results in a small prediction error
that can be encoded using only one bit (L = 2).
DM is basically a 1-bit DPCM, that is, a DPCM
that uses only two levels (L = 2) for quantization
of the [ ] [ ].
q
m k m k
In DM, we use a first-order predictor which is just a delay.
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Delta Modulation (DM)
In comparison to PCM (and DPCM), it is a very
simple and inexpensive method of A/D conversion.
A 1-bit code word in DM makes word framing
unnecessary at the transmitter and the receiver.

This strategy allows us to use fewer bits per sample
for encoding a baseband signal.
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Delta Modulation (DM)
68
69
| | | | | |
| | | | | |
| | | |

=
=
+ =
+ =
k
m
q q
q q q
q q q
m d k m
k d k m k m
k d k m k m
0

1 2 1
Hence
1
( Integrator)
( differentiator )
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Delta Modulation (DM)
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and Granular Noise
72
. ) ( of slope local the to relative large too is
size step when occurs noise granular hand, other On the

) (
max
require we , distortion overload - slope avoid To
t m
dt
t dm
Ts
A
>
A
and Granular Noise
73
Slope overload and granular noise reduce the
dynamic range of DM
frequency
Output SNR is proportional to
(For single integration case) (BT/B)^3
(For double integration case) (BT/B)^5
Comparison with PCM: at low BT/B, DM is
superior; at high BT/B, the advantage is reversed
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Comparison with PCM
Single Integration
Double Integration
Performance Comparison:
PCM Vs DPCM/DM
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Line Coding
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Digital Data Transmission
Source
Input to a digital system is in the form of sequence of digits. It could be from a
data set, computer, digitized voice signal (PCM or DM), digital camera, fax
machine, television, telemetry equipment etc.

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Line Coding and Decoding
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Data Rate Vs. Signal Rate
Data rate: the number of data elements (bits) sent in 1sec (bps). Its
also called the bit rate or transmission rate.
Signal rate: the number of signal elements sent in 1sec. Its also
called the pulse rate, the modulation rate, symbol rate or the baud
rate.
Transmission bandwidth is related to baud rate.
We wish to:
increase the data rate (increase the speed of
transmission)
decrease the signal rate (decrease the bandwidth
requirement)
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Line Codes
Output of the transmitter is coded into
electrical pulses or waveforms for the
purposes of transmission over the channel or
to modulate a carrier.
This process is called line coding or
transmission coding.
There are many possible ways to assign a
waveform (pulse) to a digital data based of
various desirables.
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Line coding schemes
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1 1
1
1 1 1 0 0 0
t
1 is encoded with p(t) and 0 is encoded with no pulse.
Pulse returns to zero level after every 1.
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1 is encoded with p(t) and 0 is encoded with p(t).
Pulses returns to zero level after every 1 and 0.
1 1
1
1 1 1
0 0
0
t
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1 is encoded with p(t) or p(t) depending on
whether previous 1 is encoded p(t) or p(t)
while 0 is encoded with no pulse.
Pulses returns to zero level after every 1 and 0.
Also known as Pseudoternary or Alternate Mark Inversion (AMI)
1 1
1
1
1
1
0 0
0
t
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1 is encoded with p(t)
while 0 is encoded with no pulse.
Pulses do not return to zero level after every 1 and 0.
1 1
1
1 1
1
0 0
0
t
85
1 is encoded with p(t)
while 0 is encoded with p(t).
Pulses do not returns to zero level after every 1 and 0.
1 1
1
1 1
1
0 0
0
t
86
Desirable Properties of Line Codes
Transmission bandwidth
Power efficiency
Error detection and correction capability
Favorable power spectral density
Transparency
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Desirable Properties
of Line Codes
Transmission bandwidth
It should be as small as possible.
Power efficiency
For a given bandwidth and specified detection
error probability, transmitted power should be as
small as possible.
Error detection and correction capability
It should be possible to detect and if possible to
correct detected errors.
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Desirable Properties
of Line Codes
Favorable power spectral density
It is desirable to have zero PSD at e=0 (dc) as ac
coupling and transformers are used at the repeaters.
It should be possible to extract timing or clock
information from the signal.
Transparency
It should be possible to transmit a digital signal
correctly regardless of the pattern of 1s and 0s.
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PSD of Various Line Codes:
Assumptions
Pulses are spaced Tb seconds apart. Consequently, the
transmission rate is Rb=1/ Tb pulses per second.
The basic pulse used is denoted by p(t) and its Fourier
transform is P(e).
The PSD of the line code depends upon that of the pulse
shape p(t). We assume p(t) to be a rectangular pulse of width
Tb/2 i.e.

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PSD of Polar Signaling
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Polar Signaling
Essential bandwidth of the signal is 2R
b
Hz.
This is four times the theoretical BW (Nyquist)
Polar signaling has no error detection
capability.
It has non-zero PSD at e=0.
Polar signaling is the most power-efficient
scheme.
Transparent

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93
PSD On-Off Signaling
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On-Off Signaling
For a given transmitted power, it is less immune to noise interference
than polar scheme.
Made up of a polar signal plus periodic signal; hence, BW is similar to
polar signaling (Fig 7.2. Page 296, Lathi).
Contains a discrete component of clock frequency (Eq 7.19, Lathi).
PSD of On-Off signaling is of that of polar signaling (Eq 7.19, Lathi).
Non-transparent.
All the disadvantages of polar schemes such as:
Excessive transmission bandwidth
Non-zero power spectrum at e=0
No error detection capability.
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PSD of Bipolar Signaling
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(Pseudoternary or AMI) Signaling
Spectrum has DC null.
Bandwidth is not excessive
Has single error detection capability (If error
then violation of AMI rule).
If rectified, an off-on signal is formed that has
a discrete component at clock frequency.

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