Pulse Code Modulation

(PCM)

Md. Sarwar Hosen

Lecturer, Dept. of EEE, RUET
 Merits of Digital Communication:
1. Digital signals are very easy to receive. The receiver has to just detect
whether the pulse is low or high.
2. AM - FM signals become corrupted over much short distances as
compared to digital signals. In digital signals, the original signal can be
reproduced accurately.
3. The signals lose power as they travel, which is called attenuation. When
AM and FM signals are amplified, the noise also get amplified. But the
digital signals can be cleaned up to restore the quality and amplified by
the regenerators.
4. The noise may change the shape of the pulses but not the pattern of the
pulses.
5. AM and FM signals can be received by any one by suitable receiver. But
digital signals can be coded so that only the person, who is intended for,
can receive them.
6. AM and FM transmitters are ‘real time systems’. I.e. they can be received
only at the time of transmission. But digital signals can be stored at the
receiving end.
7. The digital signals can be stored, or used to produce a display on a
computer monitor or converted back into analog signal to drive a loud
speaker.
 Sampling Theorem
 1. Analysis. A band-limited signal of finite energy that has no
frequency components higher than W hertz is completely described
by specifying the values of the signal at instants of time separated by
1/2W seconds.
 2. Synthesis. A band-limited signal of finite energy that has no
frequency components higher than W hertz is completely recovered
from knowledge of its samples taken at the rate of 2W samples per
second.
 The sampling rate of 2W samples per second for a signal bandwidth
of W hertz is called the Nyquist rate; its reciprocal 1/2W (measured in
seconds) is called the Nyquist interval.
 The analysis part of the sampling theorem applies to the transmitter.
The synthesis part of the theorem, on the other hand applies to the
receiver. Note also that the Nyquist rate is the minimum sampling
rate permissible.
In this case fs>2B
In this case fs=2W
 Aliasing Effect
If the sampling rate is below Nyquist rate then this condition is known as under sampling.
Due to under sampling aliasing phenomenon occurs.
sampled at the rate of 𝒇𝒔 = 𝟏.𝟓𝒇𝒎𝒂𝒙, 𝒘𝒉𝒆𝒓𝒆 𝒇𝒎𝒂𝒙 =
Figure shows the spectrum of a message signal. The signal is

𝟏𝑯𝒛, is the maximum frequency.

 Pulse Modulation

Analog Pulse Modulation Digital Pulse Modulation

Pulse Amplitude (PAM) Pulse Code (PCM), DPCM

Pulse Width (PWM) Delta (DM), ADM

Pulse Position (PPM)

Pulse Code Modulation
• PCM is a method of converting an analog signal into a digital signal.
(A/D conversion)
• The amplitude of Analog signal can take any value over a continuous
range i.e. it can take on an infinite values.
• Digital signal amplitude can take on finite values.
• Analog signal can be converted into digital by sampling and
quantizing.
 Pulse Code Modulation (PCM):
* Analog signal is converted into digital signal by using a digital
code.
* Analog to digital converter employs two techniques:

1. Sampling: The process of generating pulses of zero width

and of amplitude equal to the instantaneous amplitude of the
analog signal. The no. of pulses per second is called
“sampling rate”.

2. Quantization: The process of dividing the maximum value

of the analog signal into a fixed no. of levels in order to
convert the PAM into a Binary Code.
The levels obtained are called “quanization levels”.

* A digital signal is described by its ‘bit rate’ whereas analog

signal is described by its ‘frequency range’.

* Bit rate = sampling rate x no. of bits / sample

 V Sampling,
o Quantization and
l Coding
t
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Time
7 111
L 6 110 B
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v 4 100 n o
e 3 011 a d
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s 1 001 s
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Time
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A PCM Generator or Transmitter
PCM Transmission Path

Block Diagram of a Repeater

PCM Receiver
What is quantization? How an analog signal is quantized? What are
the types of quantization?
The digitizing of analog signals involves the rounding off the values
which are approximately equal to the analog values. The method of
sampling chooses a few points on the analog signal and then these points
are joined to round off the value to a near stabilized value. Such a
process is called as Quantization. The following figure represents an
analog signal. The signal to get converted into digital has to undergo
sampling and quantization.
Quantization is representing the sampled values of the amplitude by a
finite set quantization level. Both sampling and quantization result in the
loss of information. The quality of a quantizer output depends upon the
number of quantization levels used. The spacing between two adjacent
representation levels is called a step size.

Types of quantization:

There are two types of quantization: a) Uniform Quantization and b)

Non uniform quantization. The type of quantization in which the
quantization levels are uniformly spaced in termed as uniform
quantization. The type of quantization in which the quantization levels
are unequal and mostly the relation between them is logarithmic, is
termed as a Non uniform quantization. There are two types of uniform
quantization. They are Mid-Rise type and Mid-Tread type. The
following figures represent the two types of uniform quantization.
The quantizer characteristic can also be of a midtread or midrise type. Figure 10.8(a) shows the input–
output characteristic of a uniform quantizer of the midtread type, which is so called because the origin lies
in the middle of a tread of the staircase like graph. Figure 10.8(b) shows the corresponding input–output
characteristic of a uniform quantizer of the midrise type, in which the origin lies in the middle of a rising
part of the staircase like graph. Note that both the midtread and midrise types of uniform quantizers,
illustrated in Fig. 5.10, are symmetric about the origin.
Working Principle of Quantizer
Quantization Noise/Error in PCM
Prove that the output signal to quantization noise ratio in decibel can be
expressed as 1.8 + 20log10L or 1.8 +6v.
 Non-uniform quantization
SNR is an indication of the quality of the received signal
Ideally we would like to have constant SNR
Unfortunately, the SNR is directly proportional to the signal power,
which varies from talker to talker
The signal power can also vary because of the connecting circuits
SNR vary even for the same talker, when the person speaks softly
Smaller amplitudes pre-dominate in speech and larger amplitude
much less frequent.
This means the SNR will be low most of the time
Cont.
• The root of this difficulty is that the quantization steps are of uniform
value
• The quantization noise is directly proportional to the square of the
step size.
• The problem can be solved by using smaller steps for smaller
amplitudes as shown in fig. on the next slide
Cont.
Cont.
• The same result can be obtained by first compressing a signal and
then using uniform quantization
• The input-output characteristics of compressor are shown in fig.
Cont.
• The horizontal axis is normalized input signal and the vertical axis is
the output signal y.
• The compressor maps the input signal into larger increments
• Hence the interval delta(m) contains large number of steps when m is
small
• The quantization noise is small for smaller input signal
• Thus loud talker and stronger signals are penalized with higher noise
steps in order to compensate the soft talker and weak signals
Compression Laws
• There are two laws regarding compressions
• (1)
• This law is used in North America and Japan

• (2) A-Law
• This law is used in Europe and the rest of the word
Cont.
• The compressed samples are restored to their original values at
receiver by using an expander
• The compressor and expander together are called compandor.
• Compression of a signal increases its bandwidth but in PCM, we are
not compressing the signal but its samples the number of samples
does not change, therefore bandwidth does not rise
• When meu-law compandor is used then output SNR is
Transmission BW and output SNR
• For binary PCM, we assign distinct group of n binary digits to each of
the L quantization levels

• Each quantized level is encoded into n-bits

• Minimum channel BW is
• This is the theoretical minimum transmission bandwidth required to
transmit the PCM signal
Example 6.2
• A signal m(t) band-limited to 3kHz is sampled at a rate 33.33% higher
than Nyquist rate, a maximum acceptable error in the sample
amplitude is 0.5% of the peak amplitude. The quantized samples are
binary coded. Find the minimum channel BW required to transmit the
coded signal. If 24 such channels are time-division multiplexed,
determine the minimum transmission BW required to transmit the
multiplexed signal
Solution
Exponential Increase of output SNR
SNR in decibel scale
Cont.
Example
6.3
Comments on Logarithmic Units
• Very small and very large values are expressed in logarithmic units
 T1 carrier system
• A schematic of T1-system is shown in fig.
Cont.
Con
t.
The T1 carrier system used in digital telephony multiplexes
24 voice channel based on 8 bit PCM. Each voice signal is
usually put through a low pass filter with cut off frequency
3.4 kHz. The filtered signal is sampled at 8 kHz. In addition a
single bit is added at the end of frame for the purpose of
synchronization.
a. The duration of each bit
b. The resultant transmission rate
c. Minimum required transmission bandwidth