Pulse Code Modulation

Why a Particular Encoding Technique

Digital data, digital signal

Equipment less complex and expensive than digital-to-
analog modulation equipment
Analog data, digital signal
Permits use of modern digital transmission and
switching equipment
 Why a Particular Encoding Technique

Digital data, analog signal

Some transmission media will only propagate analog
signals
E.g., optical fiber and unguided media
Analog data, analog signal
Analog data in electrical form can be transmitted easily
and cheaply
Done with voice transmission over voice-grade lines
Criteria For Signal Encoding
What determines how successful a receiver will be
in interpreting an incoming signal?
Signal-to-noise ratio
Data rate
Bandwidth
An increase in data rate increases bit error rate
An increase in SNR decreases bit error rate
An increase in bandwidth allows an increase in
data rate
Factors Used to Compare
Encoding Schemes
 Signal spectrum
 With lack of high-frequency components, less bandwidth required
 With no dc component, ac coupling via transformer possible
 Transfer function of a channel is worse near band edges
 Clocking
 Ease of determining beginning and end of each bit positionSignal
interference and noise immunity
 Performance in the presence of noise
 Cost and complexity
 The higher the signal rate to achieve a given data rate, the greater
the cost
Reasons for Analog Modulation
Modulation of digital signals
When only analog transmission facilities are available,
digital to analog conversion required
Modulation of analog signals
A higher frequency may be needed for effective
transmission
Modulation permits frequency division multiplexing
Basic Encoding Techniques
Analog data to analog signal
Amplitude modulation (AM)
Angle modulation
 Frequency modulation (FM)
 Phase modulation (PM)
Spectrum of AM signal
Amplitude Modulation
Transmitted power

 na 2

Pt = Pc 1 + 

 in s(t)2
 P = total transmitted power
t 
 Pc = transmitted power in carrier
Single Sideband (SSB)
Variant of AM is single sideband (SSB)
Sends only one sideband
Eliminates other sideband and carrier
Advantages
Only half the bandwidth is required
Less power is required
Disadvantages
Suppressed carrier can’t be used for synchronization
purposes
Pulse Modulation

Analogue modulated systems are quite widely used, because

of their simplicity.

An alternative to analogue modulated systems is Pulsed

systems.

This system is based on digital signals or pulses.

The basis of such a system is the use of a digital carrier
signal, which is modulated by an analogue signal.

There are various ways in which this can be achieved, giving

rise different systems.
 Sampling of signals

An analogue signal is transmitted continuously in its entire form.

This need not be done provided certain conditions are satisfied.

Samples of the analogue signal may be transmitted at given

intervals of time.

The original signal may then be recovered at the receiving end

from the transmitted samples.

This technique is known as sampling and it underlies pulsed

systems.
Consider a train of signals, with a repetition frequency f and
period T where
1
f =
T
If the pulse train were amplitude modulated by the
analogue signal, the result will be pulses whose
amplitudes are samples of the analogue signal at time
intervals T.
 If the amplitude modulated pulse train is then
analysed, its spectrum will consist of Fourier
components,

0, f , 2 f , 3 f ...
At each of this there will be a set of sum and difference frequencies
(lower and upper sidebands) due to each frequency component of
the analogue signal.
If W and f-W do not overlap then it is possible to separate the group of
frequencies at the receiving end.

This can be achieved by use of a low-pass filter with a cut-off

frequency f = W.

The separated frequencies are then those of the analogue signal

transmitted.

From this the following condition is derived

f −W ≥W f ≥ 2W
This means the repetition frequency must be at least twice the
highest frequency component in the analogue signal.

The minimum sampling frequency is then

f min = 2W
This is also called the Nyquist rate.

If the sampling rate is less than 2W the lower sideband will

overlap the baseband and it will not be possible to separate them.

This effect is known as aliasing. It can be avoided by passing the

signal through a filter before sampling.
Telephone siganls range from 300 Hz to 3.4 kHz, the internationally
agreed sampling frequency is 8kHz. It means there is a guard band of
1.2kHz between the lower side band and the baseband.

Comment:

It means W has an upper limit

The technique can only be used if the bandwidth can be restricted to W

without destroying the essential information. To achieve this, band-
limited signals are used.
 Sampling Theorem

Any function of time F (t ) whose highest

frequency is W Hz can be completely
determined by sampled amplitudes spaced at
1
time intervals 2W apart.

If a signal f(t) is sampled at regular intervals of time

and at a rate higher than twice the highest
frequency, then the samples contain all of the
information of the original signal. The function f(t)
may be reconstructed from these samples by the use
of a low pass filter.
 Sampler

An analogue sampler commonly known as the sample-and-

hold is used in sampling the input analogue signal voltage and
maintaining that voltage until the next sampling instant.
The FET (Field Effect Transistor) acts like a simple switch.

When turned "on," it provides a low‑impedance path to

deposit the analogue sample voltage on capacitor.

The time that the FET is "on" is called the aperture or

acquisition time.

Essentially, the capacitor is the hold circuit. When the

switch FET is "Off," the capacitor does not have a complete
path to discharge through and therefore stores the sampled
voltage.
The storage time of the capacitor is also called the conversion
time because it is during this time that the unit converts the
sample voltage to a digital code.

The acquisition time should he very short. This assures that a

minimum change occurs in the analogue signal while it is being
deposited across the capacitor.

If the input to the sampler is changing while it is performing the

conversion, distortion results. This distortion is called aperture
distortion.
Thus, by having a short aperture time and keeping the input to the
constant relatively constant, the sample‑and‑hold circuit reduces
aperture distortion.

If the analogue signal is sampled for a short period of time and the
sample voltage is held at constant amplitude during the conversion
time, this is called flat‑top sampling.

If the sample time is made longer and the analogue‑to‑digital

conversion takes place with a changing analogue signal, this is called
natural sampling.

Natural sampling introduces more aperture distortion than flattop

sampling and requires a faster A/D converter.
 Pulse Code Modulation
PCM is the most commonly used technique in
digital communications
Used in many applications:
Telephone systems
Digital audio recording
CD laser disks
voice mail
digital video etc.

They are a primary building block for advanced

communication systems
Pulse Code Modulation
Based on the sampling theorem
Each analog sample is assigned a binary code
Analog samples are referred to as pulse amplitude
modulation (PAM) samples
The digital signal consists of block of n bits, where
each n-bit number is the amplitude of a PCM pulse
Quantization
Is the process of converting the
sampled signal to a binary value
Each voltage level will correspond to a
different binary number
The magnitude of the minimum step
size is called the resolution.
The error resulting from quantizing is
called the quantization noise. Its value
is 1/2 the resolution
Pulse Code Modulation
 Dynamic Range

This is the ratio of the largest to smallest analogue

signal that can be transmitted.
Vmax
DR =
Vmin
But Vmin is the resolution and can be written as
Vmax
q = Vmin = n
2
It follows that
Vmax
DR = = 2n
Vmin
If this is expressed in decibels
Vmax
DR(dB) = 20 log = 20 log 2n = 20n log 2 = 6.02n
Vmin

DR(dB) = 6n

From DR = 2n It can be observed that the DR is the

Maximum binary number for a system. With one

code used for 0V which is not considered in
calculating DR, it is observed that
DR = 2n − 1
 Example

Given a PCM system with the following parameters:

Maximum analog input frequency 3kHz
A maximum decoded voltage at the receiver of +/- 1.27V
A minimum dynamic range of 35dB.
Find the minimum sample rate
The number of bits required
The resolution
The quantization error
Reasons for Growth of Digital
Techniques
Growth in popularity of digital techniques for sending
analog data
Repeaters are used instead of amplifiers
 No additive noise, can be used over long distances
TDM is used instead of FDM
 No intermodulation noise
 For a given bandwidth the signal/noise ratio is superior
Conversion to digital signaling allows use of more
efficient digital switching techniques
 Fits in well with other digital systems