Chapter 3.

Baseband Pulse and Digital Signaling

Chapter Objec:ves

Ø Analog-to-digital signaling

Ø Binary and mul:level digital signals

Ø Spectra and bandwidths of digital signals

Ø Preven:on of intersymbol interference

Ø Time division mul:plexing

Ø Packet transmission
 3.1. Introduc:on
The following are the four main goals of this chapter
² To study how analog waveforms can be converted to digital waveform.
The most popular technique is called pulse code modula,on (PCM).

² To study how to compute the spectrum for digital signals.

² To examine how the filtering of pulse signals affects our ability to

recover the digital informa?on at the receiver. This filtering can
produce what is called intersymbol interference (ISI) in the recovered
data signal.
² To study how we can mul?plex (combine) data from several digital bit
streams into one high-speed digital stream for transmission over a
digital system. One such technique, called ,me-division mul,plexing
(TDM), will be studied in this chapter.
 3.2. Pulse Amplitude Modula:on
² Pulse amplitude modula,on (PAM) is an engineering term that is used
to describe the conversion of the analog signal to a pulse-type signal
in which the amplitude of the pulse denotes the analog informa?on.

² The purpose of PAM signaling is to provide another waveform that

looks like pulses, yet contains the informa?on that was present in
the analog waveform.
² There are two classes of PAM signals:
ü PAM that uses Natural Sampling (Ga:ng)
ü PAM that uses Instantaneous Sampling (Flat-Top PAM)
 3.2. Pulse Amplitude Modula:on
Natural Sampling (Ga:ng)

DEFINIATION: If w(t) is an analog waveform bandlimited to B hertz, the

PAM signal that uses natural sampling is:

ws (t) = w(t)s(t)

∞
t − kTs
where s(t) = ∑ Π( ) is a rectangular wave
k=−∞
τ
switching waveform and fs = 1 / Ts ≥ 2B
3.2. Pulse Amplitude Modula:on
Natural Sampling (Ga:ng)

ws(t) =w(t)s(t)
 3.2. Pulse Amplitude Modula:on
Natural Sampling (Ga:ng)

THEOREM: The spectrum for a naturally sampled PAM signal is

∞
sin π nd
Ws ( f ) = ℑ[ws (t)] = d ∑ W ( f − nfs )
n=−∞
π nd

where fs = 1 / Ts , ws = 2π fs, the duty cycle of s(t) is

d = τ / Ts , and W ( f ) = ℑ [ w(t)] is the spectrum of the

original unsampled waveform.

 3.2. Pulse Amplitude Modula:on
Natural Sampling (Ga:ng)
Natural Sampling (Ga:ng)

• The duty cycle of the switching

waveform is d = τ/Ts = 1/3.
• The sampling rate is fs = 4B.
 3.2. Pulse Amplitude Modula:on
Genera:on of PAM with natural sampling (ga:ng)
The PAM waveform with natural sampling can be generated using a
CMOS (complementary metal-oxide-semiconductor) circuit consis?ng of
A clock and analog switch as shown.
 3.2. Pulse Amplitude Modula:on
Recovering Naturally Sampled PAM
² At the receiver, the original analog waveform, w(t) can be recovered
from the PAM signal, ws(t), by passing the PAM signal through a
low pass filter, where the cutoff frequency is: B < fcutoff < fs-B .

² If the analog signal is under sampled fs < 2B, the effect of spectral
overlapping is called Aliasing. This results in a recovered analog signal
that is distorted compared to the original waveform
3.2. Pulse Amplitude Modula:on
Recovering Naturally Sampled PAM
3.2. Pulse Amplitude Modula:on
Instantaneous Sampling (Flat-Top PAM)
 3.2. Pulse Amplitude Modula:on
Instantaneous Sampling (Flat-Top PAM)

DEFINIATION: If w(t) is an analog waveform bandlimited to B hertz, the

PAM signal that uses instantaneous sampling is:

∞
ws (t) = ∑ w(kTs )h(t − kTs )
k=−∞
(
"t% * 1, | t |< τ / 2
where h(t) = ∏ $ ' = ) is sampling pulse
#τ & * 0, | t |> τ / 2
+
shape, were τ ≤ Ts = 1 / fs and fs ≥ 2B
 3.2. Pulse Amplitude Modula:on
Instantaneous Sampling (Flat-Top PAM)

THEOREM: The spectrum for a flat-top PAM signal is

∞
1
Ws ( f ) = H ( f ) ∑ W ( f − kfs )
Ts k=−∞
" sin πτ f %
H ( f ) = ℑ[h(t)] = τ $ '
# πτ f &

² This type of PAM signal consists of instantaneous samples

² w(t) is sampled at t = kTs
² The sample values w(kTs) determine the amplitude of the flat-top
rectangular pulses
3.2. Pulse Amplitude Modula:on
Instantaneous Sampling (Flat-Top PAM)
 3.2. Pulse Amplitude Modula:on
Limita:ons of PAM

² The transmission of either naturally of instantaneously sampled

PAM over a channel requires a very wide frequency response.
² The bandwidth required is much larger than that of the original
analog signal.
² The noise performance of the PAM system can never be beSer than
achieved by transmiTng the analog signal directly.
² PAM is not very good for long-distance transmission.
 3.3. Pulse Code Modula:on
DEFINITION: Pulse code modula5on (PCM) is essen?ally analog-to-digital
conversion of a special type where the informa?on contained
in the instantaneous samples of an analog signal is represented
by digital words in a serial bit stream.
ADVANTAGES:
² Rela?vely inexpensive digital circuitry may be used extensively.
² PCM signals derived from all types of analog sources may be merged
with data signals and transmiSed over a common high-speed digital
communica?on system.
² In long-distance digital telephone systems requiring repeaters, a clean
PCM waveform can be regenerated at the output of each repeater,
where the input consists of a noisy PCM waveform.
² The noise performance of a digital system can be superior to that of an
analog system.
² The probability of error for the system output can be reduced even
further by the use of appropriate coding techniques.
 3.3. Pulse Code Modula:on
PCM signal is generated by carrying out three basic opera5ons:

1 2 3

sampling quan:zing encoding

3.3. Pulse Code Modula:on
1. Sampling (Flat-top PAM Signal)
 3.3. Pulse Code Modula:on
2. Quan:zing

Example of M = 8 = 23 uniform quan:zer

M = 8 (M = 2n) levels are used to approximate the analog sample values

(infinite number):
-7, -5, -3, -1, 1, 3, 5, 7
 3.3. Pulse Code Modula:on
2. Quan:zing

The PAM signal is 5 7 7 77

“rounding off” to 5
get quan?zed
PAM signal

errors are introduced because of the finite number of levels (M = 8) is used

in the quan?zing (quan,zing noise)
 3.3. Pulse Code Modula:on
3. Encoding
The PCM signal is obtained from the quan?zed PAM signal by encoding
each quan?zed sample value into a digital word.
Three –bit M = 8 Gray code

-7 -5 -3 -1 1 3 5 7

010 011 001 000 100 101 111 110

5 7 7 7 7 5
 3.3. Pulse Code Modula:on
Prac:cal PCM Circuits
² Three popular techniques are used to implement the analog-to-
digital converter (ADC) encoding opera?on:
1. The coun?ng or ramp, ( Maxim ICL7126 ADC)
2. Serial or successive approxima?on, (AD 570)
3. Parallel or flash encoders. ( CA3318)

² The objec?ve of these circuits is to generate the PCM word.

² Parallel digital output obtained (from one of the above techniques)

needs to be serialized before sending over a 2-wire channel

² This is accomplished by parallel-to-serial converters [Serial Input-

Output (SIO) chip]

² UART,USRT and USART are examples for SIO’s

 3.3. Pulse Code Modula:on
Bandwidth of PCM Signals
² The bandwidth of PAM signal can be obtained as a func?on of the
spectrum of the input analog signal (Linear func?on).

² The bandwidth of PCM signal is not directly related to the spectrum

of input analogy signal. (Nonlinear func?on: 3 steps)

² The bandwidth of (serial) binary PCM waveforms depends on the bit

rate R and the waveform pulse shape used to represent the data.

² The bit rate R is:

R = nfs
where n is the number of bits in the PCM word (M = 2n) and fs is the
sampling rate (PAM).
 3.3. Pulse Code Modula:on
Bandwidth of PCM Signals
² For no aliasing case ( fs >=2B, B is the bandwidth of the analog
signal, that is to be converted to PCM signal ), the MINIMUM
bandwidth of PCM is:
BPCM = R/2=nfs/2
The Minimum bandwidth of nfs/2 is obtained only when sin(x)/x
pulse is used to generate the PCM waveform.

² For PCM waveform generated by rectangular pulses, the First-null

bandwidth is:
BPCM = R = nfs
² For a reasonable value of n, the bandwidth of the serial PCM signal
will be significantly larger than the bandwidth of the corresponding
analog signal that it represents.
² If the bandwidth of the PCM signal is reduced by improper filtering
of poor frequency response, one bit will smear into adjacent bit
slot. This is called intersymbol interference (ISI).
 3.3. Pulse Code Modula:on
Effects of Noise

The analog signal that is recovered at the PCM system output is

corrupted by noise. Two main effects produce this noise or distor?on:

² Quan?zing noise that is caused by the M-step quan?zer at the

PCM transmiSer.

² Bit errors in the recovered PCM signal. The bit errors are caused by
channel noise, as well as improper channel filtering, which causes ISI
(intersymbol interference).

The input analog signal needs to be sufficiently bandlimited (with a low-

pass an?aliasing filter) and sampled fast enough so that the aliasing noise
on the recovered analog signal is negligible.
 3.3. Pulse Code Modula:on
Effects of Noise

The ra:o of the recovered analog PEAK signal power to the total average
noise power is given by:

!S$ 3M 2
# & =
" N % pkout 1+ 4(M 2 −1)Pe

The ra:o of the AVERAGE signal power to the average noise power is:

!S$ M2
# & =
" N % pkout 1+ 4(M 2 −1)Pe

Where M is the number of quan?zed levels used in the PCM system and
Pe is the probability of bit error in the recovered binary PCM signal at
the receiver DAC before it is converted back into an analog signal
 3.3. Pulse Code Modula:on
Example 3-3. Average signal-to-noise ra:o for a recovered analog signal

Calculate the average SNRdB of the analog signal that is recovered from a
PCM signal that has error bits with a probability of error of Pe. Plot the
SNRdB for Pe over a range from 10-7 to 10-1.

!S$ M2
# & =
" N % pkout 1+ 4(M 2 −1)Pe

M = 2n

n=6
 3.3. Pulse Code Modula:on
If Pe is negligible, there are no bit errors resul?ng from channel noise and
no ISI, the peak SNR resul?ng from only quan?zing error is

!S$ 3M 2 !S$
# & = # & = 3M 2
" N % pkout 1+ 4(M 2 −1)Pe " N % pkout

The ra:o of the AVERAGE signal power to the average noise power is:

!S$ M2 !S$ 2
# & = # & =M
" N %out 1+ 4(M 2 −1)Pe " N %out

Above equa?ons can be expressed in decibels as:

!S$
# & = 6.02n + α
Where, M = 2n
" N %dB α = 4.77 for peak SNR
α = 0 for average SNR
 3.3. Pulse Code Modula:on
Example 3-4. Design of a PCM signal for telephone systems
Assume that an analog audio voice-frequency(VF) telephone signal occupies a band from
300 to 3,400Hz. The signal is to be converted to a PCM signal for transmission over a
digital telephone system. The minimum sampling frequency is 2x3.4 = 6.8 ksample/sec.
To be able to use of a low-cost low-pass an?aliasing filter, the VF signal is oversampled
with a sampling frequency of 8ksamples/sec. This is the standard adopted by the Unites
States telephone industry. Assume that each sample values is represented by 8 bits; then
the bit rate of the binary PCM signal is:
R = (fssamples/s)(n bits/sample)
= (8k samples/s)(8 bits/sample) = 64 k bits/s
This 64k bits/s is called a DS-0 signal (digital signal, type zero)
The minimum absolute bandwidth of the binary PCM signal is
R nf s
BPCM ≥ = = 32kHZ
2 2
This bandwidth is realized when a (sinx)/x pulse shape is used.
 3.3. Pulse Code Modula:on
Example 3-4. Design of a PCM signal for telephone systems

If a rectangular pulse for sampling, the absolute bandwidth is infinity,

and the first null bandwidth is :

BPCM = R = 64kHZ
We require a bandwidth of 64k HZ to transmit this digital voice PCM
signal, whereas the bandwidth of the original analog voice signal
was, at most, 4k Hz.
We observe that the peak signal-to-quan?zing noise power ra?o is:
!S$
# & = 3(28 ) 2 = 52.9dB
" N % pkout
 3.3. Pulse Code Modula:on
Nonuniform Quan:za:on
Many signals (e.g. voice analog) have a non-uniform distribu?on
-- The amplitude is more likely to be close to zero than to be at higher levels
-- Ex. If the peak value allowed is 1 V, weak passages my have voltage levels
on the order of 01. V (20 dB down)

6
Output sample
XQ
4

2 Example: Nonuniform 3 bit quantizer

-8 -6 -4 -2 2 4 6 8

-2
Input sample
X
-4

-6
 3.3. Pulse Code Modula:on
Nonuniform Quan:za:on

Analog signal Compression PCM circuit

w1(t) (nonlinear) (uniform quan?zer)

μ-law type: w2 (t) =

(
ln 1+ µ w1(t) )
ln(1+ µ )

In the United States, Canada, and Japan, the telephone companies

use a μ = 255 compression characteris?c in their PCM systems.
 3.3. Pulse Code Modula:on
Nonuniform Quan:za:on

M = 8 Quan?zer Characteris?c μ-law characteris?c

 3.3. Pulse Code Modula:on
Nonuniform Quan:za:on

Analog signal Compression PCM circuit

w1(t) (nonlinear) (uniform quan?zer)

"
$ A w1(t) 1
0 ≤ w1(t) ≤
$ 1+ ln A A
A-law type: w2 (t) = #
$ 1+ ln( A w1(t) ) 1
$ ≤ w1(t) ≤ 1
% 1+ ln A A

In Europe, the A-law is generally used.

 3.3. Pulse Code Modula:on
Nonuniform Quan:za:on

The Output SNR follows the 6-dB law

!S$
# & = 6.02n + α
" N %dB

Where α = 4.77 - 10log (V/xrms) (uniform quan?zing)

α ≈ 4.77 - 10log [ln(1+μ)] (μ-law quan?zing)

α ≈ 4.77 - 10log [1 + lnA] (A-law quan?zing)