Lesson 2 –

Formatting and baseband modulation

Nathaniel Nelson
ENG304 - Communication II Nathaniel Nelson
 Baseband systems

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 Formatting and transmission
of baseband signals.

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 Formatting of textual data
(Character coding)

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 Seven-bit American standard code for
information interchange

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 Extended Binary Coded Decimal
Interchange Code

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 Example 1
You want to transmit the word “HOW” using an 8-ary system. Encode
the word “HOW” into a sequence of bits, using 7-bit ASCII coding,
followed by an eighth bit for error detection, per character. The eighth
bit is chosen so that the number of ones in the 8 bits is an even number.
i. How many total bits are there in the message?
ii. Partition the bit stream into k = 3 bit segments. Represent each of
the 3-bit segments as an octal number (symbol). How many octal
symbols are there in the message?
iii. If the system were designed with 16-ary modulation, how many
symbols would be used to represent the word “HOW"?
iv. If the system were designed with 256-ary modulation, how many
symbols would be used to represent the word “HOW”?

ENG304 - Communication II Nathaniel Nelson

 Example 2
We want to transmit 800 characters/s, where
each character is represented by its 7-bit ASCII
codeword, followed by an eighth bit for error
detection, per character, as in Problem 2.1. A
multilevel PAM waveform with M = 16 levels is
used.
a. What is the effective transmitted bit rate?
b. What is the symbol rate?

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 Example 3
We wish to transmit a 100-character alphanumeric message
in 2 s, using 7-bit ASCII coding, followed by an eighth bit
for error detection, per character, as in Problem 2.1. A
multilevel PAM waveform with M = 32 levels is used.
a. Calculate the effective transmitted bit rate and the
symbol rate.
b. Repeat part (a) for 16-level PAM, eight-level PAM.
four-level PAM and PCM (binary) waveforms.

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 MESSAGES, CHARACTERS,
AND SYMBOLS

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 FORMATTING ANALOG
INFORMATION
❖If the information is analog, it cannot be character
encoded as in the case of textual data; the information
must first be transformed into a digital format.
❖The process of transforming an analog waveform into
a form that is compatible with a digital
communication system starts with sampling the
waveform to produce a discrete pulse-amplitude-
modulated waveform.

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 The Sampling Theorem
❖ The process of transforming an analog waveform into a
form that is compatible with a digital communication
system starts with sampling the waveform to produce a
discrete pulse-amplitude-modulated waveform.
❖ The sampling theorem states that: A bandlimited signal
having no spectral components above fm hertz can be
determined uniquely by values sampled at uniform
intervals of
1
Ts  sec
2 fm

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 Cont’d
❖This particular statement is also known as the uniform
sampling theorem.
❖Stated another way, the upper limit on T, can be
expressed in terms of the sampling rate, denoted
𝑓𝑠 = 1 𝑇 𝑠 . The restriction, stated in terms of the
sampling rate, is known as the Nyquist criterion. The
statement is
fs  2 fm

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 Example 4
An analog signal is sampled at its Nyquist rate
1ITS, and quantized using L quantization levels.
The derived digital signal is then transmitted on
some channel.
a. Show that the time duration, T, of one bit of
the transmitted binary encoded signal must
satisfy T< T's/(log2L.).
b. When is the equality sign valid?

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 Example 5
Determine the minimum sampling rate necessary
to sample and perfectly reconstruct the signal
x(t) = sin (6280f)/(6280f).

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 Impulse Sampling
❖ The validity of the sampling theorem using the frequency
convolution property of the 𝑓𝑠 = 1 𝑇 𝑠 transform is
demonstrated.
❖ Assume an analog waveform, x(f), with a Fourier transform,
X(f), which is zero outside the interval −𝑓 𝑚 < 𝑓< 𝑓𝑚
❖ The sampling of x(t) can be viewed as the product of x(f)
with a periodic train of unit impulse functions 𝑥𝛿(𝑡) and
defined as
∞
𝑥𝛿(𝑡) = 𝛿(𝑡 − 𝑛𝑇𝑠)
𝑛−∞

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 Cont’d
❖ The sifting property of the impulse function states that
𝑥 𝑡 𝛿 𝑡− 𝑡0 = 𝑥(𝑡0)𝛿(𝑡 − 𝑡0)
❖ Using this property, we can see that xs(t) the sampled
version of x(t), is given by
∞

𝑥𝑠 𝑡 = 𝑥 𝑡 𝑥𝛿 𝑡 = 𝑥 𝑡 𝛿 𝑡− 𝑛𝑇𝑠
𝑛=−∞
∞

= 𝑥 𝑛𝑇𝑠 𝛿(𝑡− 𝑛𝑇𝑠)

𝑛=−∞

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 Cont’d
❖ Using the frequency convolution property of the Fourier
Transform, we can transform the time domain product
𝑥(𝑡)𝑥𝛿(𝑡) to the frequency domain convolution 𝑋 𝑓 𝑋𝛿 𝑓
❖ The transform 𝑋𝑠 𝑓 becomes
𝑛=∞
1
𝑋𝑠 𝑓 = 𝑋 𝑓 ∗ 𝑋𝛿 𝑓 = 𝑋 𝑓 ∗ 𝛿(𝑓 − 𝑛𝑓𝑠)
𝑇𝑠
𝑛=−∞
𝑛=∞
1
= 𝑋(𝑓 − 𝑛𝑓𝑠)
𝑇𝑠
𝑛=−∞

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 Natural Sampling

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 Example 6
❖ Given an analog waveform that has been sampled at its Nyquist
rate,𝑓𝑠, using natural sampling, prove that a waveform (proportional
to the original waveform) can be recovered from the samples, using
the recovery techniques shown in Figure P2.1. The parameter mfs is
the frequency of the local oscillator, where m is an integer.

Figure P2.1

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 Sample-and-Hold Operation
❖It can be described by the convolution of the sampled
pulse train, [ 𝑥(𝑡)𝑥𝛿 𝑡 ], with a unity amplitude
rectangular pulse p(t) of pulse width Ts. This time,
convolution results in the flattop sampled sequence

𝑥𝑠 𝑡 = 𝑝 𝑡 ∗ 𝑥 𝑡 𝑥𝛿 𝑡
𝑛=∞

= 𝑝 𝑡 ∗ 𝛿(𝑡− 𝑛𝑇𝑠)
𝑛=−∞

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 Cont’d
❖ The Fourier transform, X,(f), of the time convolution is the
frequency-domain product of the transform P(f) of the
rectangular pulse and the periodic spectrum, of the impulse-
sampled data:
𝑛=∞

𝑥𝑠 𝑓 = 𝑃(𝑓)𝐹 𝑥(𝑡) 𝛿(𝑡− 𝑛𝑇𝑠)

𝑛− ∞
𝑛= ∞
1
= 𝑃 𝑓 𝑋 𝑓 ∗ 𝛿(𝑡− 𝑛 𝑇𝑠)
𝑇𝑠
𝑛−∞
𝑛= ∞
1
= 𝑃(𝑓) 𝑋(𝑓 − 𝑛𝑓𝑠)
𝑇𝑠
𝑛−∞

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 Aliasing

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 Aliasing in the frequency domain a) continuous signal spectrum b)Sampled
signal spectrum

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 Higher sampling rates eliminates aliasing a) continuous signal spectrum
b)Sampled signal spectrum

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 Sharper cut-off filters eliminates aliasing a) continuous signal spectrum
b) Sampled signal spectrum

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 Without Oversampling
I. The signal passes through a high performance
analog lowpass filter to limit its bandwidth.
II. The filtered signal is sampled at the Nyquist rate
for the (approximated) bandlimiled signal.A
strictly bandlimited signal is not realizable.
III. The samples are processed by an analog-to-
digital converter that maps the continuous-
valued samples to a finite list of discrete output
levels.

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 With Oversampling
I. The signal is passed through a low performance (less costly) analog
low-pass filter (prefilter) to limit its bandwidth.
II. The pre-filtered signal is sampled at the (now higher) Nyquist rate
for the (approximated) bandlimited signal.
III. The samples are processed by an analog-to-digital converter that
maps the continuous-valued samples to a finite list of discrete
output levels.
IV. The digital samples are then processed by a high performance
digital filter to reduce the bandwidth of the digital samples.
V. The sample rate at the output of the digital filter is reduced in
proportion to the bandwidth reduction obtained by this digital filter.

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 SOURCES OF CORRUPTION

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 Sampling and Quantizing Effects
I. Quantization Noise
II. Quantizer Saturation
III. Timing Jitter

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 Quantization Noise
❖ The distortion inherent in quantization is a round-off or
truncation error.
❖ The process of encoding the PAM signal into a quantized
PAM signal involves discarding some of the original
analog information.
❖ This distortion, introduced by the need to approximate the
analog waveform with quantized samples, is referred to
as quantization noise.
❖ The amount of such noise is inversely proportional to the
number of levels employed in the quantization process.

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 Quantizer Saturation
❖The quantizer (or analog-to-digital converter)
allocates L levels to the task of approximating the
continuous range of inputs with a finite set of outputs.
❖The range of inputs for which the difference between
the input and output is small is called the operating
range of the converter.
❖If the input exceeds this range, the difference between
the input and the output becomes large, and we say
that the converter is operating in saturation.

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 Timing Jitter
❖ The sampling theorem predicted precise reconstruction of
the signal based on uniformly spaced samples of the signal.
❖ If there is a slight jitter in the position of the sample, the
sampling is no longer uniform.
❖ The effect of the jitter is equivalent to frequency modulation
(FM) of the baseband signal.
❖ If the jitter is random, a low-level wideband spectral
contribution is induced whose properties are very close to
those of the quantizing noise. If the jitter exhibits periodic
components, the periodic FM will induce low-level spectral
lines in the data

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 Channel Effects
I. Channel Noise
II. Intersymbol Interference

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 Channel Noise
❖ Thermal noise, interference from other users, and
interference from circuit switching transients can cause
errors in detecting the pulses carrying the digitized
samples.
❖ Channel-induced errors can degrade the reconstructed
signal quality quite quickly.
❖ If the channel noise is small, there will be no problem
detecting the presence of the waveforms
❖ On the other hand, if the channel noise is large enough to
affect our ability to detect the waveforms, the resulting
detection error causes reconstruction errors.

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 Intersymbol Interference
❖ The channel is always bandlimited. A bandlimited
channel disperses or spreads a pulse waveform passing
through it.
❖ When the channel bandwidth is much greater than the
pulse bandwidth, the spreading of the pulse will be slight.
❖ When the channel bandwidth is close to the signal
bandwidth, the spreading will exceed a symbol duration
and cause signal pulses to overlap.
❖ This overlapping is called intersymbol interference (ISI).

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 PULSE CODE MODULATION
❖Pulse code modulation (PCM) is the name
given to the class of baseband signals obtained
from the quantized PAM signals by encoding
each quantized sample into a digital word.
❖ The source information is sampled and
quantized to one of L levels; then each
quantized sample is digitally encoded into an l-
bit (l = Iog2 L) codeword.

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 Example 7
Determine the number of quantization levels that
are implied if the number of bits per sample in a
given PCM code is
(a) 5; (b) 8; (c) x.

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 BASEBAND TRANSMISSION

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 Waveform Representation of Binary
Digits

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 PCM Waveform Types
I. Nonreturn-to-zero (NRZ)
II. Return-to-zero (RZ)
III. Phase encoded
IV. Multilevel binary

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 Cont’d
❖In choosing a PCM waveform for a particular
application, some of the parameters worth examining
are as ff:
I. DC component. Eliminating the dc energy from the
signal's power spectrum enables the system to be ac
coupled. Magnetic recording systems, or systems
using transformer coupling, have little sensitivity to
very low frequency signal components. Thus low-
frequency information could be lost.

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 b) Self-Clocking
❖Symbol or bit synchronization is required for any
digital communication system.
❖Some PCM coding schemes have inherent
synchronizing or clocking features that aid in the
recovery of the clock signal.
❖For example, the Manchester code has a transition in
the middle of every bit interval whether a one or a
zero is being sent.
❖This guaranteed transition provides a clocking signal.

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 c) Error detection
❖Some schemes, such as duobinary, provide the means
of detecting data errors without introducing additional
error-detection bits into the data sequence.

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 d) Bandwidth compression
❖Some schemes, such as multilevel codes,
increase the efficiency of bandwidth utilization
by allowing a reduction in required bandwidth
for a given data rate; thus there is more
information transmitted per unit bandwidth.

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 e) Differential encoding
❖This technique is useful because it allows the polarity
of differentially encoded waveforms to be inverted
without affecting the data detection.
❖In communication systems where waveforms
sometimes experience inversion, this is a great
advantage.

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 f) Noise immunity
❖The various PCM waveform types can be
further characterized by probability of bit error
versus signal-to-noise ratio.
❖Some of the schemes are more immune than
others to noise.
❖For example, the NRZ waveforms have better
error performance than does the unipolar RZ
waveform

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 PCM Word Size
❖ For digital telephone channels, each speech sample is PCM
encoded using 8 bits, yielding 2 𝑛 or 256 levels per sample.
❖ The choice of the number of levels, or bits per sample,
depends on how much quantization distortion we are willing
to tolerate with the PCM format.
❖ It is useful to develop a general relationship between the
required number of bits per analog sample (the PCM word
size), and the allowable quantization distortion.
❖ The bits per PCM word is given as:
1
𝑙≥ log 2 𝑏𝑖𝑡𝑠
2𝑝

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 M-ary Pulse-Modulation
Waveforms
❖There are three basic ways to modulate
information on to a sequence of pulses: we can
vary the pulse's amplitude, position, or
duration, which leads to the names:
I. pulse-amplitude modulation (PAM),
II. pulse-position modulation (PPM), and
III. pulse-duration modulation (PDM)

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Please Any Questions
HAVE A NICE
DAY