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SKGOCHHAYAT,SDE,RTTCBHUBANESWAR
PULSE CODE MODULATION (PCM)
DEFINITION: Pulse code modulation (PCM) is essentially analog-
to-digital conversion of a special type where the information
contained in the instantaneous samples of an analog signal is
represented by digital words in a serial bit stream.
Basic Steps For PCM System
Filtering
Sampling
Quantization
Encoding
Line Coding
FILTERING Filters are used to limit the speech signal to the
frequency band 300-3400 Hz.

Analog to Digital Conversion
The Analog-to-digital Converter (ADC)
performs three functions:
Sampling
Makes the signal discrete in time.
If the analog input has a bandwidth
of W Hz, then the minimum sample
frequency such that the signal can
be reconstructed without distortion.
Quantization
Makes the signal discrete in
amplitude.
Round off to one of q discrete levels.
Encode
Maps the quantized values to digital
words that are 8 bits long.
If the (Nyquist) Sampling Theorem is
satisfied, then only quantization introduces
distortion to the system.
ADC
Sample
Quantize
Analog
Input
Signal
Encode
111
110
101
100
011
010
001
000
Digital Output
Signal
111 111 001 010 011 111 011
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SAMPLING PROCESS
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4.3 Sampling Theorem
X
Digital signal
s(t)
m
s
(t) m(t)
m(t)
t
m
s
(t)
t
s(t)
t
T
s
Sampling theorem states that :"If a band limited
signal is sampled at regular intervals of time and
at a rate equal to or more than twice the highest
signal frequency in the band, then the sample
contains all the information of the original
signal."
fS 2fm
Fourier series for impulse train :


6
S(t) = T
S
+ 2T
S
(Cos 2 (t T
S
)+ Cos 2x2 (t T
S
+.)
SAMPLING & COMBINING CHANNELS
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PAM OUTPUT SIGNALS
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RECONSTRUCTION OF ORIGINAL SIGNAL
9
f
3f

2
s
2f
s
3f
s
f
s 0
f
s
-f
m
f
s
+f
m 2f
s
+f
m
3f
s
+f
m
3f
s
-f
m
2f
s
-f
m
m
s
(f)
Spectrum of the sampled signal
The spectrum of the sampled signal has sidebands f
s
f
m
, 2f
s
f
m
, 3f
s
f
m
and so
on.
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The choice of sampling frequency, f
s
must follow the sampling theorem to overcome
the problem of aliasing and loss of information
(a) Sampling frequency=> f
s1
<2f
m (max)
f
2f
s1
3f
s1
f
s1
f
m
Aliasing
m
s
(f)
(b) Sampling frequency=> f
s2
>2f
m (max)
f
2f
s2
3f
s2
f
s2
f
m
m
s
(f)
Shannon sampling
theorem=>f
s
2f
m

Nyquist frequency
f
s
=2f
m
=f
N
A bandlimited signal that
has a maximum
frequency, f
max
can be
regenerated from the
sampled signal if it is
sampled at a rate of at
least 2f
max .

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Pulse Code Modulation (PCM)

Codec technique
Voice Bandwidth =
300 Hz to 3400 Hz
Sampling Stage Analog Audio Source
= Sample
8 kHz (8,000 Samples/Sec)
Nyquists Theorem says
sample at twice the bandwidth of
the line.
Voice bandwidth ~ 3400 Hz.
So, must sampling rate should be
6800 samples/sec.
PCM actually uses 8000
samples/sec since cutoff not
sharp.
Height of sampled signal above /
below the base line is converted
to a binary value
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4.4 Detection of Sampled Signal
By using LPF to the sampled signal, m
s
(t)
LPF m
s
(t) m(t)
Cut-off frequency , f
o
for LPF must be within the range: f
m
f
o
f
s
- f
m
Eventhough the sampled signal can be detected easily at f
s
=2f
m ,
but usually
f
s
>2f
m
. The main reason is to have a guardband .

Therefore, the maximum frequency that can be processed by the sampled
data using sampling frequency, f
s
(without aliasing) is:
=>f
m
= f
s
/ 2 =1 / 2T
s
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4.3.1 Difference in Sampling Methods
In every sampling methods, the pulse amplitude is directly proportional to the
amplitude of the information signal
Practically, an ideal sampling is difficult to generate
However, by using an ideal and natural sampling, noise can be eliminated, which
is not the case for flat-top sampling
I deal Sampling Flat-top Sampling
m
s
(t)
t
Natural Sampling
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Natural Sampling Flat-top Sampling
Information signal
Pulse signal
Sampled signal (PAM)
t
m(t)
t
s(t)
T
s

t
m
s
(t)
T
s

t
m
s
(t)
T
s

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16
PULSE AMPLITUDE MODULATED SIGNAL
NATURAL TOP
SAMPLING
CLOCK
The FET is the switch used as a sampling gate.

When the FET is on, the analog voltage is shorted to ground; when off,
the FET is essentially open, so that the analog signal sample appears at
the output.

Op-amp 1 is a noninverting amplifier that isolates the analog input
channel from the switching function.
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FLAT-TOP SAMPLING
SAMPLED & HOLD CIRCUIT
HIGH FANOUT
OP
AMP-2
clock

As seen in Figure, the instantaneous amplitude
of the analog (voice) signal is held as a constant
charge on a capacitor for the duration of the
sampling period T
s
.
Op-amp 2 is a high input-impedance voltage
follower capable of driving low-impedance loads
(high fanout).
The resistor R is used to limit the output current
of op-amp 1 when the FET is on and provides
a voltage division with r
d
of the FET. (r
d
, the
drain-to-source resistance, is low but not zero)
sample-and-hold circuit.
Eeng 360 19
Quantization
The output of a sampler is still continuous in amplitude.
Each sample can take on any value e.g. 3.752, 0.001, etc.
The number of possible values is infinite.
To transmit as a digital signal we must restrict the number of
possible values.
Quantization is the process of rounding off a sample according to
some rule.
E.g. suppose we must round to the nearest tenth, then:
3.752 --> 3.8 0.001 --> 0

QUANTIZING-POSITIVE SIGNAL
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QUANTIZING - SIGNAL WITH + Ve & - Ve VALUES
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Quantization Interval
Represent the voltage value for each quantized level
For example: For a sampled signal that has 5V amplitude, V
pp

= 10 V divide by the quantized level, L

= 8 level,
Therefore, quantized interval , qi=10V/8=1.25V
Quantization level, L = 2
n
Quantization level depends on the number of binary bits, n
used to represent each sample.
For example:For = 3; Quantization level, L = 2
3
= 8 level.
In this example, first level (level 0) is represented by 000,
whereas bit 111 represents the eigth level
3 terms that are commonly used in the quantization
process:
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Quantization value, V
k

The middle voltage for each quantized level
For example: for n = 3, quantized level, L

= 8 and a sampled
sinusoidal signal with +5 V ,
The middle quantized value for level 0,
V = - 5V+(1.25V2) = - 4.375V

In this example, for a sample that is in level 0 segment will
be represented by bit 000 with a voltage value of 4.375 V.
The difference between the sampled value and the
quantized value results in quantization noise.
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For voice communication 256 levels are commonly
used (i.e n = 8)
Quantization
1. Quantizing operation approximates the analog
values by using a finite number of levels. This
operation is considered in 3 steps
a) Uniform Quantizer
b) Quantization Error
c) Quantized PAM signal output
2. PCM signal is obtained from the quantized PAM
signal by encoding each quantized sample value
into a digital word.
Uniform Quantization
Most ADCs use uniform
quantizers.
The quantization levels of a
uniform quantizer are
equally spaced apart.
Uniform quantizers are
optimal when the input
distribution is uniform.
When all values within the
Dynamic Range of the
quantizer are equally likely.
Input sample X
Example: Uniform 3 bit quantizer
q=8 and X
Q
= {1,3,5,7}
2 4 6 8
1
5
3
Output sample
X
Q

-2 -4 -6 -8
Dynamic Range:
(-8, 8)
7
-7
-3
-5
-1
Quantization Characteristic
t
Level 0 : 000
Level 1 : 001
Level 2 : 010
Level 3 : 011
Level 4 : 100
Level 5 : 101
Level 6 : 110
Leve l 7 : 111
1.9V
+5.0V
-5.0V
4.375V
3.125V
1.875V
0.625V
-0.625V
-1.875V
-3.125V
-4.375V
4.3V
1.9V
-3.2V
-4.5V
Quantization level &
binary representation
Quantized
value
Sampled signal
UNIFORM QUANTIZATION
Uniform quantization is a quantization process with a uniform (fixed) quantization
interval.
Example : n = 3 , L

= 8 , signal +5 V ; => V
k
= 1.25 V . Bit rate: f
b
= n
f
s

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Uniform Quantization using Folded Binary Code (sign bit)
+m
p
-m
p
0
1 11
1 10
1 01
1 00
0 00
001
0 10
0 11

value
Sign bit
t
100 101 111 111 111 110 101 000 010 011 011 010 000 001 101 110 110 101
Quantization error
Q
e
PCM code
t
The same code representing several
samples with different amplitudes
Step size
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28
1000
0111
0110
0101
0100
0011
0010
0001
1001
1010
1011
1100
1101
1110
1111
A brief aside about ADCs
1000 1110 1111 1011 0100 0001 0011
Numbers passed from ADC to computer to represent analogue voltage
ADCs are used to convert an analogue input voltage into a number that can
be interpreted as a physical parameter by a computer.
Resolution=
1 part in 2
n
Quantization error
May add to or substract from the
actual signal
Quantization error (Q
e
) is also called Quantization noise (Q
n
) .
And its maximum magnitude is one half of the voltage of the
minimum step size .
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Quantization error
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Input voltage range: 14 mV to
+14 mV

Binary
number

Input voltage
range (mV)

1 11

10 to 14

1 10

6 to 10

1 01

2 to 6

1 00

0 to 2

0 00

-2 to 0

0 01

-6 to -2

0 10

-10 to -6

0 11

-14 to -10

Example : Uniform Quantization error
Q
n
= LSB voltage /2 = q
i
/2
14 mV = 28 mV with 8 steps and 8 codes.
Therefore Q
n
= 28/8 = 3.5 mV.
Therefore : Q
n
= 3.5 mV / 2 = 1.75 mV
SNR
q
= [1.76 + 6.02n] dB
Noise from quantization error
can be reduced by increasing
the quantization level i.e
increase n.
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Companding
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Pulse Code Modulation - Analog to Digital Conversion
Stage 1
Quantizing Stage
Output

100100111011001
A-Law (Europe)
-Law (USAJapan)
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2 Popular companding system (standardized by ITU)
EUROPE => A - Law
USA/NORTH AMERICA => - Law
A
x for
x
A
for
A
Ax
A
Ax
y
1
0
1
1
log 1
log 1
) log( 1
( (
( (

+
+
+
=
A - compressor paramater. Usually the
value of A is 87.6.
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USA/NORTH AMERICA => - Law
Law is a standard compress-
expand that is used in America and
Japan. The value of used is 255 (8
bit).
( )

+
+
=
1 log
) 1 log( x
y
(max) i
i
E
E
x =
(max) o
o
E
E
y =
For both laws, the values of x and y
refers to the equation below:
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ENCODING
CURVE WITH
COMPRESSION
8 BIT CODE
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P A B C W X Y Z
Sign bit SEGMENT
1 FOR POSITIVE &
0 FOR NEATIVE

0 0 0 0 0 0 0
1 1 1 1 1 1 1
Example : PCM-TDM CEPT System
Frame structure and Timing : European standard PCM system : E Line
(a) bits per time slot (b) time slots per frame (c) frames per multiframe
488 ns
3.9 s
3.9 s
125 s
125 s
2 ms
8 bits per
time slot
Bit duration
30 signal + 2 control = 32 channels = 1 frame
Signalling & synchronization
16 frames = 1 multiframe
Duration of multiframe
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PCM30 basic frame
1 2 3 4 5 6 7 8
X 1
Y Y Y
1

1

1

Bit No.
Value
Message
ca. 3,9 s
1 2 3 4 5 6 7 8
Bit numbering
32 x 8 = 256 bit
125 s
t
voice
1
voice
2
voice
15
voice
16
voice
30
0 1 2 15 16 17 31
SIG
1 2 3 4 5 6 7 8
0 0 1 1 0 1 1
Bit No.
value
Frame alignment
X
PDH E1 signal
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Bit 1 X Used in international connections
Bit 3 Y=1 FRAME SYNCHRONISATION
Bit 4 Y=1 HIGH ERROR DENSITY
Bit 3,4,5 111 Urgent alarm
Bit 6-8 111 Reserved for national options
Bit 1, X Used in
international
connections,
FAW :-0011011

Frame structure and timing
Number of channel = 32
Number of bits in one time slot = 8
32 channels = 1 frame
Number of bits in a frame = 32 x 8 = 256 bits
PCM 32 channels (30 signals + 2 control)
This frame must be transmitted within the sampling period and thus 8
x 10
3
frames are transmitted per second.
Therefore :
Transmission rate = 8 x 10
3
x 256 = 2.048 Mb/s
Bit duration = 1 / 2.048 x 10
6
= 488 ns
Duration of a time slot = 8 x 488 ns = 3.9 s
Duration of a frame = 32 x 3.9 s = 125 s => (= 1 / 8 kHz = 125 s)
Duration of a multi frame = 16 x 125 s = 2 ms
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Europe bit rate(Mb/s)

2.048

8.448

34.368

139.264

565.148

Telephone
channel
30

120

480

1920

7680

SDH

2.5Gb/s

Telephone
channel

North America bit
rate(Mb/s)
24

1.544

48

3.152

96

6.321

672

44.736

4032

274.176

Bit rate for PCM & Higher Order Mux
European standard : A-Law
30 + 2 control channel = 32
Bit rate= 32 x 8 bit/sample x 8000 sample/s
= 2.048 Mb/s
North American standard (NAS) : -Law
For every 24 sample, 1 bit is added for
synchronization
For 24 sample => 24 x 8 bit/sample + 1 bit
= 193 bits
Bit rate= 193 x 8000 = 1.544 Mb/s
Needs Multiplexing Process of transmitting two or more
signals simultaneously
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LINE CODE
24 CHL PCM AMI

30 CHL PCM HDB-3

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PCM Transmission System
SKGOCHHAYAT,SDE,RTTCBHUBANESWAR
The advantages of PCM are:

Relatively inexpensive digital circuitry may be used
extensively.
PCM signals derived from all types of analog sources
may be merged with data signals and transmitted over
a common high-speed digital communication system.
In long-distance digital telephone systems requiring
repeaters, a clean PCM waveform can be regenerated
at the output of each repeater, where the input consists
of a noisy PCM waveform.
The noise performance of a digital system can be
superior to that of an analog system.
The probability of error for the system output can be
reduced even further by the use of appropriate coding
techniques.
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skgochhayat@gmail.com 45