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PULSE CODE MODULATION

and
Delta Modulation

1 Bandwidth

2 Data Formats

3 Quantization

4 Transmission of PCM signals

5 Delta Modulation
Bandwidth
Important to know the bandwidth (BW) of a signal . The
signal BW decides the BW of amplifiers , filters and channel.
Yet there is no one acceptable definition of BW .

2
signal s(t) S(f) , S(f) = Gss ( f ) = spectrum

Gss ( f ) Gss ( f )

f f
0 fc 0 fl fc fu
Baseband or Lowpass Bandpass. It is also narrowband
if f c >> f u f l
One definition: half power BW.
This is the interval between half power points, i.e.,
3 db below the peak value.

Gss ( f )

A
0.5 A

f
0 fl fu
Another definition is null-to-null

Gss ( f )

T
t 1 1 f

T 0 0
2 2 T T
1
BW = , T = Pulse width
T
Gss ( f )

f
t
T 0 1 fc 1
T ( fc ) ( fc + )
2 T T
2
2
BW = Hz
fc T
Exercise

p (t )

t
T T

2 2

What is the relationship between BW and pulse width ?

Data Stream
Assume original message is alphanumeric. The information must
be encoded before transmission. Standard encoding formats are:
American Standard Code for Information Interchange (ASCII)
Extended Binary Coded Decimal Interchange Code (EBCDIC)

Message (text) T
H N
I K
character encoding
6 bit ASCII 010
001 000
100
100
100
011
100
110
100

8-ary digit 1 2 0 4 4 4 3 4 6 4
(symbols)

8-ary Waveform S1 (t ) S2 (t ) S0 (t ) S4 (t ) S4 (t ) S4 (t ) S3 (t ) S4 (t ) S6 (t ) S4 (t )
Suppose we use 32-ary digits

T H I N K

0
00101 000100 100100 011100 110100

32-ary digits 5 1 4 17 25 20
(symbols)

Waveform

8-ary requires 10 off (8-ary symbols)

32-ary requires 6 off (32-ary symbols)
Quantization
Easier to transmit samples of signals if we use pulses
of equal heights. Output of A/D is a pulse amplitude
modulated (PAM) signal. Quantization converts all voltages
to one of L = 2n levels. These levels are encoded to give
pulse code modulation (PCM) signals.

s(t)
5 +5 +3 3 2
3
Quantizer 1 1 0 1 1 0 1 1 0 0 1 100 10
A/D 4 bit
3 2

PAM
Flat Top PCM
sampling
Vp
2V p
Suppose quantizer has L = 128 = 2 levels
7

X L
(i.e. n = 7) ,
and PAM signal limited to V p volts.

X 2V p
L=128 levels
Then each quantization step q = L
v.

q q
Quantization error e volts
2 2

Vp
Suppose s(t) has uniform distribution between V p . Then
f (e)
1
q

q
e
q 0

2 2
q
2
2
e2 q2 1 2V p
Variance of e is q q
de =
12
=
12
L

2

Ps = E {s 2 (t )}
Ps
Quantization SNR = 3L2 2
Vp
SNR increases with L or number of bits, n , in fact

SNR = 1010 22 + 1010 3
2
= + . if Ps Vp2
Tutorial
The dynamic range (DR) of a system is the ratio of the maximum input
power to the minimum input power that the system can process
without distortion. Thus, the minimum power must be above the
systems own noise level. In db, the DR is
Peak Signal Power
10 log
Noise Power
For a quantizer that has Vp peak, bits, the Q noise power is
2
Vp
, L = 2 .
3L2 2
Hence the DR is Vp
10 log 2
Vp
3L2
= 10 log 3 + 10 log 2 2 4.8 + 6

Thus DR increases by 6 db / bit .

A high DR is important for digital audio and images to provide
high clarity and resolution.

The CD player has 16 bits and a DR of 96 db

Exercise
Find the number of bits required for an A/D to give a DR of 80.
If the input signal has a maximum voltage of 5 V , how many
volts does the LSB represent ?
Nonuniform Quantization
Vp
Speech amplitudes mostly lie below 2 . Only 15% of speech
samples will exceed.
For the same number of bits, can have higher quantization
SNR if we use nonuniform quantization: more levels at
low voltages and less levels at high voltages.

output

input
Better to use uniform quantizer, therefore it is common to
compress the analog signals to produce same effects as
nonuniform quantization.
output,y The compressor amplifies low input
voltage and attenuates the high, called
compression. After quantization, an
x,input inverse compression, called
expansion, restores the compressed
signal.
s(t)
uniform
A/D D/A
Quantizer

compressor Expander

The whole process is called companding.

- law Compressor
North America y = ymax
[
ln 1 +
x
xmax
]sgn x
ln[1 + ]

+1 for x 0
= 255 and sgn x =
1 for x < 0

Asia and Europe A x x

x 1
use y max
max 1 + ln A sgn x , 0 <
A-law xmax A
y=
1 + ln xA x
ymax max 1 x
A = 87.6 sgn x , < <1
1 + ln A A xmax
ymax
Note that for = 0 , y = x
xmax
Pulse Code Modulation

xq
Reconstructed
waveform
Transmission of PCM Signals
The T-1 Carrier System
1 8 KHz LPF 1
LPF

coder
8 bits/channel
decoder LPF 2
24
LPF

LPF 24

channel Add one synchronizing bit

at end of each frame.
1 2 24 1
PCM
8 bit
RZ AMI

125 sec
One Frame
Notes. 1. An 8-bit codeword from each channel. The first
bit is the sign bit. The last 7 bits give the magnitude
of the sample.
2. A frame contains 8 bits from each channel, plus a
synchronizing bit at the end of a frame. Every 12
frames form a 12-bit synchronizing bit pattern which
the receiver detects to decide if there is synchronization.
3. There are 8 x 24+1=193 bits per frame or per 125 sec
since sampling at 8 KHz.
193
4. Bit rate is 6
= 1.544 Mbits / sec
125 10
5. For signaling purposes, a signaling bit is put in to
replace the LSB of each channel. This occurs once
every 6 frames.
Historical Notes
1. The T-1 carrier PCM system was an important product for AT&T
in the 1960s. Its main use was for phone service between high
density cities of less than 100 miles distance. Timing jitters
prohibit using T-1 for longer distances.

2. Repeaters at every 6000 ft. restore the PCM pulse shapes. This
permits the transmission of 1.544 Mbps over phone lines of only
3KHz BW. This is an examples of compensating small BW by high
SNR.

3. The T-1 pulse format is RZ-AMI . The signal has no dc

component and the repeaters can be transformer coupled.
Exercise

3. This BW is >> 3KHz BW of a standard phone line. Why can we

use phone lines to transmit the T-1 PCM signal ?
Differential Pulse Code Modulation
Wish to reduce the transmission rate (bits/sec) so that the
required BW can be smaller. Can reduce the number of bits
to represent a digit, i.e., the number of quantization levels L.
But quantization Ps
SNR = 3L2
Vp 2

reduce Vp.

Suppose we can predict s(n) from s(n-1), s(n-2) , i.e.

s(n) = a1s (n 1) + a2 s (n 2)
Instead of transmitting s(n), we transmit
d (n) = s (n) s(n)
The receiver will reconstruct using own prediction and
d(n) to get
s (n) = d (n) + s(n) .
If d(n) is small, can maintain same SNR by having a
smaller L, because Vp is now also smaller.

Problem is s (n), s(n) are not quantized. Need to

quantize d(n) before transmission.

Quantizing d q ( n ) = d ( n ) + e( n )

quantization error
Transmit d q (n)
s (n) + d (n)
Quantizer d q (n)

+
DPCM +
Transmitter
s(n) ~
s ( n)
Predictor

+
d q (n) ~
s ( n)
+

s(n)
Predictor

s(n) = a1~
s (n 1) + a2 ~
s (n 2)
~
s (n) = d q (n) + s(n) = s (n) + e(n)
Example. In speech, a second order predictor is
s(n) = a1s (n 1) + a2 s (n 2) , a1 = 1.1314 , a2 = 0.3714

d (n) = s (n) s(n)

= s (n) a1s (n 1) a2 s (n 2)

{ }
Pd = E d 2 (n) = (1 + a1 + a2 ) Ps + (2a1a2 2a1 ) R1 2a2 R2
2 2

=R1 E {s (n) s (n=

1)} 0.825 Ps
From measurements

=R2 E {s (n) s (n=
2)} 0.562 Ps
on speech
{
Ps = E s 2 (n) }
Pd = 0.2753 Ps
Exercise
Speech signals approximately follow the equation
S (n) = a1 S (n 1) + a2 S (n 2)

Thus we can predict S(n) from S(n-1) and S(n-2) using

S (n) = a1 S (n 1) + a2 S (n 2)

Suppose we transmit
d (n) = S (n) - S (n)

instead of S(n). Show how the receiver produce the speech signals.
Calculate the reduction in transmitting power when sending d(n)
instead of S(n). Given : E{S(n)S(n - 1)} = 0.8
E{S(n)S(n - 2)} = 0.5
Delta Modulation
The closer are the samples s(n), s(n-1), s(n-2), the easier it
is to perform linear prediction. DM oversamples by 4
Nyquist rate and encodes signals using only 1-bit (2 level).
Thus DM is a 1-bit DPCM.

1-bit quantizer
s (n) ~
+ ~ s (t )
+
s (t ) d q (n)
LPF d q (n) s ( n)
LPF
+

+
+
Delay by T
~
s ( n)
Delay by T Receiver
a
Transmitter a
Delta Modulation
The closer are the samples s(n), s(n-1), s(n-2), the easier it
is to perform linear prediction. DM oversamples by 4
Nyquist rate and encodes signals using only 1-bit (2 level).
Thus DM is a 1-bit DPCM.

1-bit quantizer
s (n) ~
+ ~ s (t )
+
s (t ) d q (n)
LPF d q (n) s ( n)
LPF
+

+
+
Delay by T
~
s ( n)
Delay by T Receiver
a
Transmitter
a
A practical implementation is

comparator
s (t ) + d (t ) d q (n) s(t ) ~
s (t )
a LPF

s(t )
a

integrator
Transmitter

The transmitter compares s(t) with s(t ) and the sign of the
error is multiplied by , sampled at 4 Nyquist rate,
and sent.
d q (n)is a train of narrow pulses of height either + or .
The integrator sums up ( integrates along time) the pulses,
multiplies by a, to produce s(t ) to try to follow s(t). This
is similar to a feedback control system.

Start up Slope overload

0 t
s(t )
d q (n) s (t )

0 t

d (t )

0 t
s(t ) is a stair-case approximation of s(t) . The receiver
passes s(t ) through a LPF to give a smooth approximation
to s(t). d (n) s(n) s(n 1) , which is the derivative of s(n),
hence the name DM.
2
R(1)
1 1
R ( 0)
a =
The optimum gain R(1)
R(0)

R (1) = E{s (t ) S (t + 1)} , { }

R (0) = E s 2 (t )

Granular noise occurs when s(t ) is tracking s(t) closely,

d(n) is just a sequence of + and . Can reduce
granular noise by reducing .
Overload occurs when s(t) changes too fast and s(t ) too
slow to catch up. This happens when slope of s (t ) > ,
2T
T=sampling time.
Can prevent overload with large and small T. But large
produces large granular noise and T requires high sampling.

Detects overload by sensing a long sequence of + or
pulses. When this occurs, increase . Detects granular noise
by a sequence of alternating + and pulses. When this occurs,
decrease .
Communication between space shuttle and ground control
used DM because of its simplicity.
Summary
1. PCM quantizes a digital number to one of L = 2l
levels and encodes it to be an l-bit word. Encoding
pulses are of equal amplitudes.
2. There are many formats to represent a binary sequence.
The choice of a particular format is dependent on
application.
3. Quantization error is a source of noise. It decreases
with increasing number of bits used.
4. The T-1 carrier is the first PCM system for speech
transmission.
5. DPCM can reduce the bit rate for the same SNR.
6. DM is a form of 1-bit DPCM. Its main advantage is
simplicity.
References
1. Digital Communications, B.Sklar, PTR Prentice Hall, 1988

3. Modern Digital and Analog Communication Systems,

B.P. Lathi, 3rd Edition, Oxford University Press, 1998