Digital Transmission

Digital Communications
Engr. Rose Anne Reano, PUP-STB
 Objectives

■ Define digital transmission

■ List and describe the advantages and
disadvantages of digital transmission
■ Briefly describe pulse width modulation, pulse
position modulation, and pulse amplitude
modulation
■ Define and describe pulse code modulation
■ Explain flat-top and natural sampling
 Objectives

■ Describe the Nyquist sampling theorem

■ Describe folded binary codes
■ Define and explain dynamic range
■ Explain PCM coding efficiency
 Objectives

■ Describe signal-to-quantization noise ratio

■ Explain the difference between linear and
nonlinear PCM codes
■ Describe idle channel noise
 Objectives

■ Describe idle channel noise

■ Explain several common coding methods
■ Define companding and explain analog and
digital companding
■ Define digital compression
Digital Transmission
Digital Modulation
 Digital Transmission

Digital transmission is the transmittal of

digital signals between two or more points in
a communications system.
 Digital Transmission

• 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
 Advantages of Digital
Transmission
1. Noise immunity
2. Ease of processing
3. Simpler to measure and evaluate
 Disadvantages of Digital
Transmission
1. Requires greater bandwidth
2. Needs additional circuitry for
conversion from analog to digital
or vice versa
3. Incompatible with older analog
systems
Pulse Modulation
Digital Modulation
 Pulse Modulation

• Pulse modulation consists essentially of

sampling analog information signals and
then converting those samples into
discrete pulses and transporting the
pulses from a source to a destination over
a physical transmission medium.
 Pulse Modulation

Four methods of PM:

1.pulse width modulation (PWM)
2.pulse position modulation (PPM)
3.pulse amplitude modulation (PAM)
4.pulse code modulation (PCM)
 Pulse Modulation

• 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.
Pulse Modulation
 Pulse Modulation

• 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.
Pulse Modulation
 Pulse Modulation

• With PAM, the amplitude of a constant

width, constant-position pulse is varied
according to the amplitude of the sample
of the analog signal.
Pulse Modulation
 Pulse Modulation

• With PCM, the analog signal is sampled

and then converted to a serial n-bit binary
code for transmission.

• Each code has the same number of bits

and requires the same length of time for
transmission.
Pulse Modulation
 Pulse Modulation

• PAM is used as an intermediate form of

modulation with PSK, QAM, and PCM,
although it is seldom used by itself.
 Pulse Modulation

• PWM and PPM are used in special-

purpose communications systems mainly
for the military but are seldom used for
commercial digital transmission systems.
 Pulse Modulation

• PCM is by far the most prevalent form of

pulse modulation and, consequently, will
be discussed in more detail.
Pulse Code Modulation
Digital Modulation
 Pulse Code Modulation

• Alex H. Reeves is credited with inventing

PCM in 1937 while working for AT&T at its
Paris laboratories.
 Pulse Code Modulation

• The term pulse code modulation is

somewhat of a misnomer, as it is not
really a type of modulation but rather a
form of digitally coding analog signals.
Pulse Code Modulation
 Pulse Code Modulation

• The bandpass filter limits the frequency of

the analog input signal to the standard
voice-band frequency range of 300 Hz to
3000 Hz.
 Pulse Code Modulation

• The sample-and-hold circuit periodically

samples the analog input signal and
converts those samples to a multilevel
PAM signal.
 Pulse Code Modulation

• The analog-to-digital converter (ADC)

converts the PAM samples to parallel
PCM codes,
 Pulse Code Modulation

• which are converted to serial binary data

in the parallel-to-serial converter and then
outputted onto the transmission line as
serial digital pulses.
 Pulse Code Modulation

• The transmission line repeaters are

placed at prescribed distances to
regenerate the digital pulses.
 Pulse Code Modulation

• In the receiver, the serial-to-parallel

converter converts serial pulses received
from the transmission line to parallel PCM
codes.
 Pulse Code Modulation

• The digital-to-analog converter (DAC)

converts the parallel PCM codes to
multilevel PAM signals
 Pulse Code Modulation

• The hold circuit is basically a lowpass filter

that converts the PAM signals back to its
original analog form.
 PCM Sampling

• 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.
 PCM Sampling

Two basic techniques used to perform the

sampling function:
1.Natural sampling
2.Flat-top sampling
 PCM 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.
PCM Sampling
 PCM Sampling

• The most common method used for

sampling voice signals in PCM systems is
flattop sampling, which is accomplished in
a sample-and-hold circuit.
 PCM Sampling

• With flat-top sampling, the input voltage is

sampled with a narrow pulse and then
held relatively constant until the next
sample is taken.
PCM Sampling
 PCM Sampling

• As the figure shows, the sampling process

alters the frequency spectrum and
introduces an error called aperture error,
which is when the amplitude of the
sampled signal changes during the
sample pulse time.
 PCM Sampling

• Schematic diagram of a sample-and-hold

circuit.
 PCM Sampling

• The FET acts as a simple analog switch.

When turned on, Q1 provides a low-
impedance path to deposit the analog
sample voltage across capacitor C1.
 PCM Sampling

• The time that Q1 is on is called the

aperture or acquisition time.
 PCM Sampling

• Essentially, C1 is the hold circuit. When

Q1 is off, C1 does not have a complete
path to discharge through and, therefore,
stores the sampled voltage.
 PCM Sampling

• The storage time of the capacitor is called

the A/D conversion time because it is
during this time that the ADC converts the
sample voltage to a PCM code.
 PCM Sampling

• The acquisition time should be very short

to ensure that a minimum change occurs
in the analog signal while it is being
deposited across C1.
 PCM Sampling

• If the input to the ADC is changing while it

is performing the conversion, aperture
distortion results.
 PCM Sampling

• Thus, by having a short aperture time and

keeping the input to the ADC relatively
constant, the sample-and-hold circuit can
reduce aperture distortion.
 PCM Sampling

• Figure shows the

input analog
signal, the
sampling pulse,
and the waveform
developed across
C1.
 PCM Sampling

• It is important that
the output
impedance of
voltage follower
Z1 and the on
resistance of Q1
be as small as
possible.
 PCM Sampling
This ensures that
the RC charging
time constant of the
capacitor is kept
very short, allowing
the capacitor to
charge or
discharge rapidly
during the short
acquisition time.
 PCM Sampling
• The rapid drop in
the capacitor
voltage
immediately
following each
sample pulse is
due to the
redistribution of
the charge
across C1.
 PCM Sampling
• The interelectrode
capacitance
between the gate
and drain of the
FET is placed in
series with C1 when
the FET is off, thus
acting as a
capacitive voltage-
divider network.
 PCM Sampling
• Also, note the
gradual discharge
across the capacitor
during the
conversion time.
 PCM Sampling
• This is called droop
and is caused by
the capacitor
discharging through
its own leakage
resistance and the
input impedance of
voltage follower Z2.
 PCM Sampling
• Therefore, it is
important that the
input impedance of
Z2 and the leakage
resistance of C1 be
as high as possible.
 PCM Sampling
• Essentially, voltage
followers Z1 and Z2
isolate the sample-
and-hold circuit (Q1
and C1) from the
input and output
circuitry.
 PCM Sampling

• The Nyquist sampling theorem

establishes the minimum sampling rate
(fs) that can be used for a given PCM
system.
 PCM Sampling

• For a sample to be reproduced accurately

in a PCM receiver, each cycle of the
analog input signal (fa) must be sampled
at least twice.
 PCM Sampling

• Consequently, the minimum sampling rate

is equal to twice the highest audio input
frequency.
 PCM Sampling

• If fs is less than two times fa, an

impairment called alias or foldover
distortion occurs. Mathematically, the
minimum Nyquist sampling rate is

fs ≥ 2fa
where fs minimum Nyquist sample rate (hertz)
fa maximum analog input frequency (hertz)
PCM Sampling
 PCM Sampling

• 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.
 PCM Sampling

• Using Nyquist’s sampling theorem

(Equation 1), we have fs ≥ 2fa therefore,
fs ≥ 8 kHz

• If a 5-kHz audio frequency entered the

sample-and-hold circuit, the output
spectrum shown in Figure 7 is produced.
PCM Sampling
 PCM Sampling

• It can be seen that the 5-kHz signal

produces an alias frequency of 3 kHz that
has been introduced into the original
audio spectrum.
 PCM Sampling

• The input bandpass filter shown in Figure

2 is called an antialiasing or antifoldover
filter.
 PCM Sampling

• Its upper cutoff frequency is chosen such

that no frequency greater than one-half
the sampling rate is allowed to enter the
sample-and-hold circuit, thus eliminating
the possibility of foldover distortion
occurring.
 PCM Sampling

• The codes currently used for PCM are

sign-magnitude codes, where the most
significant bit (MSB) is the sign bit and the
remaining bits are used for magnitude.
 PCM Sampling

• Table 1 shows an n-bit PCM code where

n equals 3.
 PCM Sampling

• Quantization is the process of converting

an infinite number of possibilities to a finite
number of conditions.
 Quantization

• 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.
 Quantization

• The magnitude difference between

adjacent steps is called the quantization
interval or quantum.
 Quantization

• Assigning PCM codes to absolute

magnitudes is called quantizing.

• The magnitude of a quantum is also called

the resolution.
 Quantization

• The resolution is equal to the voltage of

the minimum step size, which is equal to
the voltage of the least significant bit
(Vlsb) of the PCM code.
 Quantization

• 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.
Quantization
Quantization
Quantization
 Quantization

• For the PCM coding scheme shown in

Figure 8, determine the quantized voltage,
quantization error (Qe), and PCM code for
the analog sample voltage of 1.07 V.
Quantization
 Dynamic Range

• Dynamic range (DR) is the ratio of the

largest possible magnitude to the smallest
possible magnitude (other than 0 V) that
can be decoded by the digital-to-analog
converter in the receiver.
 Dynamic Range

• Mathematically, dynamic range is:

DR = Vmax / Vmin

DR = Vmax / resolution

where DR dynamic range (unitless ratio)

Vmin the quantum value (resolution)
Vmax the maximum voltage magnitude that can be discerned by the
DACs in the receiver
 Dynamic Range

• Dynamic range is generally expressed as

a dB value; therefore,

DR = 20 log Vmax / Vmin

 Dynamic Range

• The number of bits used for a PCM code

depends on the dynamic range.
• The relationship between dynamic range
and the number of bits in a PCM code is

2^n - 1 ≥ DR

where n number of bits in a PCM code, excluding the sign bit

DR absolute value of dynamic range
Dynamic Range
Dynamic Range
Dynamic Range
 Dynamic Range

• For a PCM system with the following parameters,

determine (a) minimum sample rate, (b) minimum
number of bits used in the PCM code, (c) resolution, and
(d) quantization error.

Maximum analog input frequency 4 kHz

Maximum decoded voltage at the receiver 2.55 V
Minimum dynamic range 46 dB
Dynamic Range
Dynamic Range
 Coding Efficiency

• Coding efficiency is the ratio of the

minimum number of bits required to
achieve a certain dynamic range to the
actual number of PCM bits used.
Coding Efficiency
Reference Book:
Advanced Electronic
Communications Systems
by Wayne Tomasi

Thank you for listening! ^_^

 Objectives

■ Describe signal-to-quantization noise ratio

■ Explain the difference between linear and
nonlinear PCM codes
■ Describe idle channel noise
 Objectives

■ Describe idle channel noise

■ Explain several common coding methods
■ Define companding and explain analog and
digital companding
■ Define digital compression