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Digital Communication

1.18 Digital Communication

1.1.5 Application of Sampling Theorem — PAM/TDM

 The Sampling Theorem allows us to replace a continuous time signal
(analog signal) by a discrete sequence of pulses in the time domain.
 One of the most significant application of the Sampling Theorem is in the
field of communication in which an analog signal is sampled by a train of
pulses, and sample values are used to modify certain parameters of a periodic
pulse train. This leads us to various analog pulse modulation techniques such
as
1. Pulse Amplitude Modulation (PAM) by varying the amplitude of the
pulses in proportion to the instantaneous value of the analog signal.
2. Pulse Width Modulation (PWM) by varying the width of the pulses in
proportion to the instantaneous value of the analog signal.
3. Pulse Position Modulation (PPM) by varying the position (starting point
on time scale) of the pulses in proportion to the instantaneous value of the
analog signal.
In all these cases, we detect the information of the pulse-modulated signal and
reconstruct the original analog signal.

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1.18 Digital Communication

1.1.5 Application of Sampling Theorem — PAM/TDM

1.18 Digital Communication

 One significant advantage of using the PAM technique is that it permits the
simultaneous transmission of several baseband signals on a time-sharing
basis,

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1.18 Digital Communication

 known as PAM—time division multiplexing (PAM-TDM).

 In fact, PAM is suitable for both TDM and digital representation of analog
signals.
 Multiplexing of several PAM signals is possible because various signals are
kept distinct and are separately recoverable because they are sampled at
different times.
 For example, Time-Division Multiplexing (TDM) systems use PAM for
multiplexing many baseband analog signals.

• The block diagram below demonstrates the PAM-TDM principle.

• In this system, 4 different signals are sampled at the same rate (
𝑓𝑠 samples per seconds).
• The clock frequency 𝑓𝐶𝐿𝐾 must be fast enough to send the all the
samples without missing a sample from a signal (in this case 𝑓𝐶𝐿𝐾
= 4𝑓𝑠 .
• The input signals are pre-filtered to prevent the aliasing.
Complete synchronization between MUX and DEMUX is critical
for correct reception.

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1.18 Digital Communication

Let us now try to understand the concept and design of PAM/TDM systems with
the help of following examples.

SOLVED EXAMPLE 1.1.16

Four analog information signals are required to be transmitted by a PAM/TDM
system. One out of four signals is band-limited to 3 kHz, whereas the remaining
three signals are band-limited to 1 kHz each. Design a TDM scheme where each
information signal is sampled at its Nyquist rate.
Solution:
 The analog signals are sampled using pulse amplitude modulation (PAM),
prior to time-division multiplexing.
 For the given data, Figure 1.1.10 shows the design of the PAM/TDM system.

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1.18 Digital Communication

Figure 1.1.10 Design of PAM/TDM System for Example 1.1.16

The multiplexer (MUX) is a single-pole rotating mechanical or electronic
switch or commutator, rotating at fs (sampling frequency) rotations per second,
such that fs ≥ 2 fm where fm is the highest signal frequency present in all the
channels.
We know that Nyquist rate, fs = 2 fm, where fm is the highest frequency present
in the analog information signal after band-limiting. Therefore,
∑ For the first band-limited signal, s1(t) of given fm1 = 3 kHz; fs1 = 6 kHz
∑ For the second band-limited signal, s2(t) of given fm2 = 1 kHz; fs2 = 2 kHz

PCM, and Delta Modulation and Demodulation 1.19

So number of poles of commutator switch connected to s1(t) = 3

Thus, total number of poles of commutator switch connected to all signals = 6
Recommended speed of the commutator = 2000 rotations per second

Number of samples per second for signal s1(t) = 3× 2000= 6000 samples/second
Number of samples per second forsignal s2(t) = 1× 2000 = 2000 samples/second
Number of samples per second forsignal s3(t) = 1× 2000 = 2000 samples/second
Number of samples per second forsignal s4(t) = 1× 2000 = 2000 samples/second
Hence, signaling rate = 6000 + 2000 + 2000 + 2000 = 12000 samples/seccond

We know that minimum transmission bandwidth, BTDM = (1/2) × signaling rate

Hence, minimum transmission bandwidth, BTDM = (1/2) × 12000 = 6000 Hz
Alternatively, minimum transmission bandwidth, BTDM = fm1 + fm2 + fm3 + fm4
Hence, minimum transmission bandwidth, BTDM = 6 kHz

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PCM, and Delta Modulation and Demodulation 1.21

1.2.1 Block Diagram of PCM

Figure 1.2.1 shows a simplified block diagram of a single-channel one-way
PCM system.

Line Speed Conversion

Clock Clock

The band-pass filter (BPF), or low-pass filter, limits the frequency of the input analog signal to
desired baseband signal frequency range.
∑ The sample-and-hold circuit periodically samples the analog input signal and converts these
samples to a multi-level PAM signal.
∑ The analog-to-digital (A/D) converter performs the function of the quantizer and encoder. Its
output is a sequence of binary symbols (also known as code words) for each sample. Each symbol
consists of a train of pulses in which each pulse may represent a binary digit.

1.22 Digital Communication

∑ PCM codes are then converted to serial binary data in the parallel-to-serial converter
and then presented to the transmission medium as serial digital pulses. Thus, the signal
transmitted over the communication channel in a PCM system is referred to as a
digitally encoded signal.
∑ When the digitally encoded signal is transmitted, noise is added during the
transmission along the channel.
∑ The transmission channel regenerative repeaters are placed at prescribed distances
to regenerate the digital pulses and enable to remove interference, if any, due to
channel noise.
∑ In the PCM receiver, the serial-to-parallel converter converts serial pulses received
from the transmission line to parallel PCM codes.
∑ The digital-to-analog (D/A) converter generates sequence of quantized multi-level
sampled pulses, resulting in reconstituted PAM signal.
∑ The hold circuit is basically a low-pass filter (LPF) to reject any frequency component
lying outside its baseband. It converts the recovered PAM signal back to its original
analog form.

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1.22 Digital Communication

• PCM belongs to a class of signal coders in which an analog signal is approximated by

mimicking the amplitude-versus-time waveform; hence known as waveform coders.
• In principle, waveform coders are designed to be signal-independent.
• PCM is used in:
1. digital telephone systems (trunk lines)
2. digital audio in computers and various compact disc formats
3. digital videos, etc.
4. PCM is the preferred method of communications within Integrated Services
Digital Network (ISDN) because with PCM it is easy to combine digitized voice
and digital data into a single, high-speed digital signal and transmit it over either
coaxial or optical fiber cables.

1.22 Digital Communication

1.2.2 PCM Sampling

Let an arbitrary analog signal s(t) be applied to a pulse modulator circuit (may be a high-
speed transistor switching circuit) controlled by a pulse signal c(t). The pulse signal
consists of an infinite succession of rectangular pulses of amplitude A, pulse width Tb,
and occurring with time interval Ts. Figure 1.2.2 depicts a simplified block diagram of
pulse sampler.

Assume that the analog signal s(t) does not contain any frequency component outside
the frequency range from –fm to +fm, and that the sampling rate fs > 2 fm (Nyquist
criterion) so that there is no aliasing. The effect of using ordinary pulses of finite duration
on the spectrum of a sampled signal is illustrated in Figure 1.2.4.

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PCM, and Delta Modulation and Demodulation 1.23

1.24 Digital Communication

1.2.3 Quantization of Sampled Signal

• The value of the sampled analog signal can be rounded off to one of the nearest
permissible numbers, known as quantized levels.
• The operation of quantization is represented in Figure 1.2.6.

Quantization Noise

At any instant of time, the difference in the values of analog signal s(t) and its
quantized signal 𝑠 (𝑡), i.e., s(t) – 𝑠 (𝑡) has a magnitude which is equal to or less than
∆/2. This difference can be regarded as noise, known as quantization noise or
quantization error.

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PCM, and Delta Modulation and Demodulation 1.25

• uniform quantization is a process in which the quantization levels are uniformly

spaced over the complete input amplitude range of the analog signal.
• That means the uniform quantizer has a linear characteristics. The largest possible
quantization noise is one-half the difference between successive levels, i.e., ±∆/2.
• the signal-to-quantization noise ratio, which is a measure of the quality of the
received signal, does not remain same over the complete input range.
• This means that the signal-to-quantization noise ratio varies with the signal level and
is higher for large signal levels.
• So it is highly desirable that signal-to-quantization noise ratio should remain
essentially constant for a wide range of input signal levels.
• The amount of quantization noise can be decreased by increasing the number of
levels, but it also increases the complexity at the receiver for decoding these levels
precisely.
• The only solution to have a constant signal-to-quantization noise ratio is to adjust the
step-size in accordance with the input signal amplitude levels. This is known as
nonuniform quantization.

PCM, and Delta Modulation and Demodulation 1.25

• Example: for uniform quantization system has :

• No. of quantization level = 8

• Peak to peak voltage of signal = 8 V
• ∴ ∆= = 1𝑉
∆
• 𝑄𝑢𝑎𝑛𝑡𝑖𝑧𝑎𝑡𝑖𝑜𝑛 𝑁𝑜𝑖𝑠𝑒 = ± = ±0.5 𝑉
• Quantization Noise Power = constant
• 𝑆𝑁𝑅 =
• But signal is not constat
• For weak signal SNR=poor

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1.26 Digital Communication

• in non-uniform quantization, the spacing between the quantization levels is not

uniform and step size varies in accordance with the relative amplitude level of the
sampled value.
• This results into an advantage of having higher average signal to quantization noise
power ratio value than that of possible in the uniform quantizer.

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1.26 Digital Communication

• the non-uniform quantization is said to be robust. In other words, robust

quantization requires that the step-size must be small for low amplitude levels and
large for high amplitude levels of analog signal. The provision for such robust
performance necessitates the use of a non-uniform quantizer.
• The non-uniform quantization technique employs an additional logarithmic amplifier
before processing the sampled speech signals by a uniform quantizer.
• The operation of a non-uniform quantizer is equivalent to passing the analog signal
through a compressor and then applying the compressed signal to a uniform quantizer
at transmitter end.
• At the receiver, a device with a characteristic complementary to the compressor,
called expander, is used to restore the signal samples to their correct relative level.
• The combination of a compressor and an expander is called a compander. So,
companding is the process of compressing the signal at transmitter end and
expanding it at the receiver end to achieve non-uniform quantization. An arrangement
of companding or robust quantization is shown in Figure 1.2.8.

PCM, and Delta Modulation and Demodulation 1.27

At Transmitter At Receiver

Figure 1.2.9 Characteristics of Compressor and Uniform Quantizer

IMPORTANT!

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PCM, and Delta Modulation and Demodulation 1.27

At Transmitter At Receiver

• ITU-T (The International Telecommunication Union) has recommended the µ-law companding
standard for use in North America and Japan.
• The compression parameters, µ determines the degree of compression.
• In the µ-law companding, the compressor characteristics are continuous, approximating a linear
dependence for low input levels and a logarithmic one for high input levels.
• The compression characteristics for µ-law (for positive amplitudes) is given by

• It is observed that the value µ = 0 corresponds to uniform quantization.

• For a relatively constant signal-to-quantization ratio and a 40 dB dynamic range, the value of
µ ≥ 100 is required (suitable for 7-bit or 128-level PCM encoding).
• An optimum value of µ = 255 has been used for all North American 8-bit or 256-level digital
terminals.

1.28 Digital Communication

ITU-T has recommended the A-law companding standard for use in Europe and rest of
the world except North America and Japan. The compression parameters, A determines
the degree of compression. The compression characteristics for A-law is given as

The value A = 1 corresponds to uniform quantization. A typical value of A = 87.6 gives

comparable results and has been standardized by the ITU-T.

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