Digital Transmission

Is the transmittal of digital signals (pulses) between two or more points in a

communications system. The signals can be binary or any other form of discrete-
level digital pulses. The original source information may be in digital form, or it
could be analog signals that have been converted to digital pulses prior to
transmission and converted back to analog signals in the receiver.

With digital transmission systems, a physical facility, such as a pair of wires,

coaxial cable, or an optical fiber cable, is required to interconnect the various
points within the system. The pulses are contained in and propagate down the
cable. Digital pulses cannot be propagated through a wireless transmission system,
such as Earth’s atmosphere or free space. AT&T developed the first digital
transmission system for the purpose of carrying digitally encoded analog signals,
such as the human voice, over metallic wire cables between telephone offices.
Today, digital transmission systems are used to carry not only digitally encoded
voice and video signals but also digital source information directly between
computers and computer networks. Digital transmission systems use both metallic
and optical fiber cables for their transmission medium.

Advantages of Digital Transmission

The primary advantage of digital transmission over analog transmission is

noise immunity. Digital signals are inherently less susceptible than analog signals
to interference caused by noise because with digital signals it is not necessary to
evaluate the precise amplitude, frequency, or phase to ascertain its logic condition.
Instead, pulses are evaluated during a precise time interval, and a simple
determination is made whether the pulse is above or below a prescribed reference
level.
Digital signals are also better suited than analog signals for processing and
combining using a technique called multiplexing. Digital signal processing (DSP)
is the processing of analog signals using digital methods and includes band
limiting the signal with filters, amplitude equalization, and phase shifting. It is
much simpler to store digital signals than analog signals, and the transmission rate
of digital signals can be easily changed to adapt to different environments and to
interface with different types of equipment. In addition, digital transmission
systems are more resistant to analog systems to additive noise because they use
signal regeneration rather than signal amplification. Noise produced in electronic
circuits is additive (i.e., it accumulates); therefore, the signal-to-noise ratio
deteriorates each time an analog signal is amplified. Consequently, the number of
circuits the signal must pass through limits the total distance analog signals can be
transported. However, digital regenerators sample noisy signals and then reproduce
an entirely new digital signal with the same signal-to-noise ratio as the original
transmitted signal. Therefore, digital signals can be transported longer distances
than analog signals.

Finally, digital signals are simpler to measure and evaluate than analog
signals. Therefore, it is easier to compare the error performance of one digital
system to another digital system. Also, with digital signals, transmission errors can
be detected and corrected more easily and more accurately than is possible with
analog signals.

Disadvantages of Digital Transmission

The transmission of digitally encoded analog signals requires significantly

more bandwidth than simply transmitting the original analog signal. Bandwidth is
one of the most important aspects of any communications system because it is
costly and limited.

Also, analog signals must be converted to digital pulses prior to transmission

and converted back to their original analog form at the receiver, thus necessitating
additional encoding and decoding circuitry. In addition, digital transmission
requires precise time synchronization between the clocks in the transmitters and
receivers.

Finally, digital transmission systems are incompatible with older analog

transmission systems.

Pulse Modulation - is a type of signal modulation where the amplitude, duration,

or position of pulses in a train of pulses is varied to represent the information being
transmitted.
 The four predominant methods of pulse modulation include pulse width
modulation (PWM), pulse position modulation (PPM), pulse amplitude modulation
(PAM), and pulse code modulation (PCM).

1. PWM is sometimes called pulse duration modulation (PDM) or pulse length

modulation (PLM), as the width (active portion of the duty cycle) of a
constant amplitude pulse is varied proportional to the amplitude of the
analog signal at the time the signal is sampled. PWM is shown in Figure 1c.
As the figure shows, the amplitude of sample 1 is lower than the amplitude
of sample 2. Thus, pulse 1 is narrower than pulse 2. The maximum analog
signal amplitude produces the widest pulse, and the minimum analog signal
amplitude produces the narrowest pulse. Note, however, that all pulses have
the same amplitude. PWM is commonly used in motor control, power
inverters, and audio applications.

2. With PPM, the position of a constant-width pulse within a prescribed time

slot is varied according to the amplitude of the sample of the analog signal.
PPM is shown in Figure 1d. As the figure shows, the higher the amplitude of
the sample, the farther to the right the pulse is positioned within the
prescribed time slot. The highest amplitude sample produces a pulse to the
far right, and the lowest amplitude sample produces a pulse to the far left.

3. With PAM, the amplitude of a constant width, constant-position pulse is

varied according to the amplitude of the sample of the analog signal. PAM is
shown in Figure 1e, where it can be seen that the amplitude of a pulse
coincides with the amplitude of the analog signal. PAM waveforms resemble
the original analog signal more than the waveforms for PWM or PPM.
 Figure 1. Pulse Modulation (a) analog signal (b) sample pulse
(c) PWM/PDM (d) PPM (e) PAM (f) PCM

PAM is used as an intermediate form of modulation with PSK, QAM, and

PCM, although it is seldom used by itself. PWM and PPM are used in special-
purpose communications systems mainly for the military but are seldom used for
commercial digital transmission systems. PCM is by far the most prevalent form of
pulse modulation.

Pulse Code Modulation (PCM) - is an encoding technique that converts analog

signal into digital signal used for digital transmission. With PCM, the pulses are of
fixed length and fixed amplitude. PCM is a binary system where a pulse or lack of
a pulse within a prescribed time slot represents either a logic 1 or a logic 0
condition. PWM, PPM, and PAM are digital but seldom binary, as a pulse does not
represent a single binary digit (bit).
Pulse Code Modulation basic building block

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

The bandpass filter limits the frequency of the analog input signal to it bandpass
frequency, allowing signal frequency within the bandpass to pass through while
rejecting the rest. The sample-and-hold circuit periodically samples the analog
input signal and converts those samples to a multilevel PAM signal. The analog-to-
digital converter (ADC) converts the PAM samples to parallel PCM codes, which
are converted to serial binary data in the parallel-to-serial converter and then
outputted onto the transmission line as serial digital pulses. The transmission line
repeaters are placed at prescribed distances to regenerate the digital pulses.

In the receiver, the serial-to-parallel converter converts serial pulses received

from the transmission line to parallel PCM codes. The digital-to-analog converter
(DAC) converts the parallel PCM codes to multilevel PAM signals. The hold
circuit is basically a lowpass filter that converts the PAM signals back to its
original analog form. An integrated circuit that performs the PCM encoding and
decoding functions is called a codec (coder/decoder).

Sample and Hold Circuit

The function of a sampling circuit in a PCM transmitter is to periodically

sample the continually changing analog input voltage and convert those samples to
a series of constant amplitude pulses that can more easily be converted to binary
PCM code. For the ADC to accurately convert a voltage to a binary code, the
voltage must be relatively constant so that the ADC can complete the conversion
before the voltage level changes. If not, the ADC would be continually attempting
to follow the changes and may never stabilize on any PCM code.

There are two basic techniques used to perform the sampling function:
Natural Sampling and Flat-top Sampling.

Natural sampling is when tops of the sample pulses retain their natural
shape during the sample interval, making it difficult for an ADC to convert the
sample to a PCM code. With natural sampling, the frequency spectrum of the
sampled output is different from that of an ideal sample. The amplitude of the
frequency components produced from narrow, finite-width sample pulses decreases
for the higher harmonics. This alters the information frequency spectrum requiring
the use of frequency equalizers (compensation filters) before recovery by a low-
pass filter.

Natural Sampling
 Flat-top sampling is the common method used for sampling voice signals
in PCM systems and is accomplished in a sample-and-hold circuit. The purpose of
a sample and hold circuit is to periodically sample the continually changing analog
input voltage and convert those samples to a series of constant-amplitude PAM
voltage levels. With flat-top sampling, the input voltage is sampled with a narrow
pulse and then held relatively constant until the next sample is taken.

Sample and Hold Circuit

Flat top Sample

Sampling - is the process of capturing and representing the amplitude of an analog
signal at discrete points in time.

Sampling Rate – is the number of samples or sample points per unit time from
which an analog signal is converted to digital form. It is governed by Nyquist
Sampling Theory which states that an analog signal must be sampled at least two
times its frequency.

fs ≥ 2fa
where:
fs = Nyquist sampling rate or frequency, samples/sec or Hz
fa = maximum analog signal frequency, Hz

Minimum Sampling Rate – the minimum sampling rate or frequency required to

convert an analog signal to digital form without distortion.

fs = 2fa

When the sampling rate is less than twice the highest analog signal
frequency, an impairment, called, alias or fold-over distortion occurs. It produces
a frequency, called the alias frequency, which is the absolute difference between
sampling frequency, fs or its multiple (2fs, 3fs, 4fs…and so on), and the analog
frequency, fa.

Sampling is a heterodyne process of mixing the analog signal with the

sample pulse in a non-linear manner. During this process, the output will generate
the original analog and sampling signal, f a and fs, their sum and difference (fs–fa,
fs+fa), and all the multiples of the fundamental frequencies, f a and fs, including their
associated cross products, like, 2fs – fa, 3fs – fa, 4fs – fa….and so on. The aliasing or
fold-over distortion can be addressed by using a bandpass filter at the input of the
PCM transmitter whose upper cut-off frequency is chosen so that no analog
frequency greater than one-half of the sampling frequency be allowed to be
sampled.
Output Spectrum of Sample and Hold Circuit (a) without aliasing (b) with aliasing

Example: For a PCM system with a maximum audio input frequency of 4 kHz,
determine the minimum sample rate and the alias frequency produced if a 5-kHz
audio signal were allowed to enter the sample-and hold circuit.

Solution:

Minimum fs = 2fa
= 2(4KHz)
= 8 KHz

The fundamental alias frequency = fs – fa

Where:
fs = 8KHz
fa = 5KHz
The fundamental alias frequency = 8KHz– 5KHz
= 3 KHz
Output Spectrum of Sample and Hold Circuit for Example Above

Quantization - is the process of converting an infinite number of possibilities to a

finite number of conditions. Analog signals contain an infinite number of
amplitude possibilities. Thus, converting an analog signal to a PCM code with a
limited number of combinations requires quantization. In essence, quantization is
the process of rounding off the amplitudes of flat-top samples to a manageable
number of levels.

For example, a sine wave with a peak amplitude of 5 V varies between 5 V

and -5 V passing through every possible amplitude in between. A PCM code could
have only eight bits, which equates to only 2 8, or 256 combinations. Obviously, to
convert samples of a sine wave to PCM requires some rounding off.

With quantization, the total voltage range is subdivided into a smaller

number of subranges. The PCM code shown in table below, is a three-bit sign-
magnitude code with eight possible combinations (four positive and four negative).
The left most bit is the sign bit (1 = + and 0 = -), and the two rightmost bits
represent magnitude. This type of code is called a folded binary code because the
codes on the bottom half of the table are a mirror image of the codes on the top
half, except for the sign bit. If the negative codes were folded over on top of the
positive codes, they would match perfectly. With a folded binary code, each
voltage level has one code assigned to it except zero volts, which has two codes,
100 (+0) and 000 (-0). The magnitude difference between adjacent steps is called
the quantization interval or quantum. For the code shown in Table 2, the
quantization interval is 1 V. Therefore, for this code, the maximum signal
magnitude that can be encoded is +3 V (111) or -3 V (011), and the minimum
signal magnitude is 1 V (101) or -1 V (001). If the magnitude of the sample
exceeds the highest quantization interval, overload distortion (also called peak
limiting) occurs.
 3-bit PCM Code

Assigning PCM codes to absolute magnitudes is called quantizing. The

magnitude of a quantum is also called the resolution. The resolution is equal to the
voltage of the minimum step size ( one step). The resolution is the minimum
voltage other than 0 V that can be decoded by the digital-to-analog converter in the
receiver. The smaller the magnitude of a quantum, the better (smaller) the
resolution and the more accurately the quantized signal will resemble the original
analog sample.

Resolution = (Vmax – Vmin)/(N-1)

where:
Vmax = Max Amplitude of Quantization Range
Vmin = Min Amplitude of Quantization Range
N = no. of steps

In the table shown above, each three-bit code has a range of input voltages
that will be converted to that code. For example, any voltage between 0.5 and 1.5
will be converted to the code 101 (1 V). Each code has a quantization range equal
to + or - the magnitude of a quantum except the codes for +0 and -0. The 0V codes
each have an input range equal to only one-half a quantum (0.5 V).