Sampling Theorem

The signals we use in the real world, such as our

voices, are called "analog" signals. To process these
signals in computers, we need to convert the signals to
"digital" form.
While an analog signal is continuous in both time
and amplitude, a digital signal is discrete in both time
and amplitude.
To convert a signal from continuous time to discrete
time, a process called sampling is used.











The value of the signal is measured at certain
intervals in time. Each measurement is referred to as a
sample.
When the continuous analog signal is sampled at a
frequency F, the resulting discrete signal has more
frequency components than did the analog signal.
If the signal contains high frequency components, we
will need to sample at a higher rate to avoid losing
information that is in the signal.
In general, to preserve the full information in the
signal, it is necessary to sample at twice the maximum
frequency of the signal. This is known as the Nyquist
rate.

What happens if we sample the signal at a frequency that is
lower that the Nyquist rate? When the signal is converted
back into a continuous time signal, it will exhibit a
phenomenon called aliasing.
Aliasing is the presence of unwanted components in the
reconstructed signal. These components were not present
when the original signal was sampled.
Aliasing occurs because signal frequencies can overlap if the
sampling frequency is too low.


Digital representations of analog waveforms
Continuous time
Continuous values
Discrete time
Discrete values
Sampling Theorem
The value of the analog signal is measured at
certain intervals in time. Each measurement is
referred to as a sample.
According to the Nyquist theorem, the
sampling rate must be
at least 2 times the highest frequency
contained in the signal.
A/D
conversion
Sampling
f(t) f
s
(t)
Introduction
rate sampling : 1
period sampling : where
(3.1) ) ( ) ( ) (
signal sampled ideal the denote ) ( Let
s s
s
s
n
s
T f
T
nT t nT g t g
t g
=
=

=
o
o
o
Frequency domain representation

=
|
|
.
|

\
|

=
n
s
s
s
T
nT t
c nT g t g sin ) ( ) (
If the sampling is at exactly the Nyquist rate, then
Under Sampling, Aliasing
Avoid Aliasing
Band-limiting signals (by filtering) before sampling.
Sampling at a rate that is greater than the Nyquist
rate.

A/D
conversion
g(t)
T
g
s
(t)
Sampling
Anti-aliasing
filter
Sampling theorem for band-limited signals of finite
energy:

1)A band-limited signal of finite energy, which has no
frequency components > W Hz, is completely described by
the samples taken at instants of time separated by 1/2W
seconds.

2) A band-limited signal of finite energy, which has no
frequency components > W Hz, may be completely
recovered from its samples taken at the rate of 2W
samples/second.

The sampling rate of 2W samples/second, for a signal
bandwidth of W Hertz, is called the Nyquist rate, and its
reciprocal 1/2W seconds is called the Nyquist interval.
The derivation of the sampling theorem is based on the
assumption that the signal g(t) is strictly band limited.
In practice, an information-bearing signal is not strictly
band limited, with some degree of under-sampling.
Consequently, some aliasing is produced by the sampling
process.

Two corrective measures for combating the aliasing
effects :
1)Prior to sampling, an anti-aliasing LPF is used to
attenuate those high-frequency components of the signal
that are not essential to the information being conveyed
by the signal.
2)The filtered signal is sampled at a rate slightly higher
than the Nyquist rate.
3.3 Pulse-Amplitude Modulation

The waveform of a PAM signal is illustrated in Figure 3.5.
The dashed curve depicts the waveform of a message signal m(t),
and the sequence of amplitude-modulated rectangular pulses shown
as solid lines represents the corresponding PAM signal s(t).

Two operations are involved in the generation of the PAM
signal, these are jointly referred to as sample and hold :

1. Instantaneous sampling of the message signal m(t) every T
s
sec,
where the sampling rate f
s
= 1/T
s
is chosen with the sampling
theorem.

2. Lengthening the duration of each sample so obtained to some
constant value T.
Pulse Amplitude Modulation
Natural Sampling
The circuit of Figure 11-3 is used to illustrate pulse
amplitude modulation (PAM). The FET is the switch used
as a sampling gate.

When the FET is on, the analog voltage is shorted to
ground; when off, the FET is essentially open, so that the
analog signal sample appears at the output.

Op-amp 1 is a noninverting amplifier that isolates the
analog input channel from the switching function.

Pulse amplitude modulator, natural sampling.
Pulse Amplitude Modulation
Natural Sampling
Op-amp 2 is a high input-impedance voltage follower
capable of driving low-impedance loads (high fanout).

The resistor R is used to limit the output current of op-amp 1
when the FET is on and provides a voltage division with r
d

of the FET. (r
d
, the drain-to-source resistance, is low but not
zero)
Pulse Amplitude Modulation
Natural Sampling
The most common technique for sampling voice in PAM
systems is to a sample-and-hold circuit.

As seen in Figure 11-4, the instantaneous amplitude of
the analog (voice) signal is held as a constant charge on a
capacitor for the duration of the sampling period T
s
.

This technique is useful for holding the sample constant
while other processing is taking place, but it alters the
frequency spectrum and introduces an error, called
aperture error, resulting in an inability to recover exactly
the original analog signal.
Pulse Amplitude Modulation
Flat-Top Sampling
The amount of error depends on how mach the
analog changes during the holding time, called
aperture time.

To estimate the maximum voltage error possible,
determine the maximum slope of the analog
signal and multiply it by the aperture time DT
Pulse Amplitude Modulation
Flat-Top Sampling
Sample-and-hold circuit and flat-top sampling.
Pulse Amplitude Modulation
Flat-Top Sampling
System for signal m(t) from PAM signal s(t)
recovering message
3.4 Other Forms of Pulse Modulation

In a pulse modulation system we may use the
increased bandwidth consumed by the pulses to improve
the noise performance of the system.
- Pulse-duration modulation (PDM), also referred to
as pulse-width modulation (PWM), where samples of
the message signal are used to vary the duration of the
individual pulses in the carrier.
- Pulse-position modulation (PPM), where the position
of a pulse relative to its un modulated time of occurrence
is varied in accordance with the message signal.


In PDM, long pulses expand considerable power while
bearing no additional information. If this unused power is
subtracted from PDM so that only time transitions are
preserved, we obtain PPM.
Accordingly, PPM is a more efficient form of pulse
modulation than PDM.

In a PPM system the transmitted information is contained
in the relative positions of the modulated pulses, the presence
of additive noise affects the performance of such a system by
falsifying the time at which the modulated pulses are judged
to occur.
Pulse width modulator.
Pulse Width and Pulse Position Modulation
Analog/pulse modulation signals.
Pulse Width and Pulse Position Modulation
Pulse position modulator.
Pulse Width and Pulse Position Modulation
PCM
PCM consists of three steps to digitize an analog
signal:
1. Sampling
2. Quantization
3. Binary encoding
Before we sample, we have to filter the signal to
limit the maximum frequency of the signal as it
affects the sampling rate.
Filtering should ensure that we do not distort the
signal, ie remove high frequency components
that affect the signal shape.
4.29
Components of PCM encoder
Sampling
Analog signal is sampled every T
S
secs.
T
s
is referred to as the sampling interval.
f
s
= 1/T
s
is called the sampling rate or sampling
frequency.
There are 3 sampling methods:
Ideal - an impulse at each sampling instant
Natural - a pulse of short width with varying amplitude
Flattop - sample and hold, like natural but with single
amplitude value
The process is referred to as pulse amplitude
modulation PAM and the outcome is a signal with
analog (non integer) values
Figure 4.24 Recovery of a sampled sine wave for different sampling rates
Quantization
Sampling results in a series of pulses of varying
amplitude values ranging between two limits: a
min and a max.
The amplitude values are infinite between the two
limits.
We need to map the infinite amplitude values onto
a finite set of known values.
This is achieved by dividing the distance between
min and max into L zones, each of height A.
A = (max - min)/L
Quantization Levels
The midpoint of each zone is assigned a
value from 0 to L-1 (resulting in L values)
Each sample falling in a zone is then
approximated to the value of the midpoint.
Quantization Zones
Assume we have a voltage signal with amplitutes
V
min
=-20V and V
max
=+20V.

We want to use L=8 quantization levels.
Zone width A = (20 - -20)/8 = 5

The 8 zones are: -20 to -15, -15 to -10, -10 to -5, -
5 to 0, 0 to +5, +5 to +10, +10 to +15, +15 to +20

The midpoints are: -17.5, -12.5, -7.5, -2.5, 2.5,
7.5, 12.5, 17.5
Assigning Codes to Zones
Each zone is then assigned a binary code.
The number of bits required to encode the zones,
or the number of bits per sample as it is commonly
referred to, is obtained as follows:
n
b
= log
2
L
Given our example, n
b
= 3
The 8 zone (or level) codes are therefore: 000,
001, 010, 011, 100, 101, 110, and 111
Assigning codes to zones:
000 will refer to zone -20 to -15
001 to zone -15 to -10, etc.
Figure 4.26 Quantization and encoding of a sampled signal
Quantization Error
When a signal is quantized, we introduce an error
- the coded signal is an approximation of the
actual amplitude value.
The difference between actual and coded value
(midpoint) is referred to as the quantization error.
The more zones, the smaller A which results in
smaller errors.
BUT, the more zones the more bits required to
encode the samples -> higher bit rate
Quantization Error and SN
Q
R
Signals with lower amplitude values will suffer
more from quantization error as the error range:
A/2, is fixed for all signal levels.
Non linear quantization is used to alleviate this
problem. Goal is to keep SN
Q
R fixed for all
sample values.
Two approaches:
The quantization levels follow a logarithmic curve.
Smaller As at lower amplitudes and larger As at higher
amplitudes.
Companding: The sample values are compressed at the
sender into logarithmic zones, and then expanded at the
receiver. The zones are fixed in height.
Bit rate and bandwidth requirements of PCM
The bit rate of a PCM signal can be calculated form the
number of bits per sample x the sampling rate
Bit rate = n
b
x f
s

The bandwidth required to transmit this signal depends on
the type of line encoding used.
A digitized signal will always need more bandwidth than
the original analog signal.
We want to digitize the human voice. What is the bit
rate, assuming 8 bits per sample?
Solution
The human voice normally contains frequencies from 0
to 4000 Hz. So the sampling rate and bit rate are
calculated as follows:
Example 4.14


The basic elements of a PCM system.
PCM Decoder
To recover an analog signal from a digitized signal
we follow the following steps:
We use a hold circuit that holds the amplitude value of
a pulse till the next pulse arrives.
We pass this signal through a low pass filter with a
cutoff frequency that is equal to the highest frequency
in the pre-sampled signal.
The higher the value of L, the less distorted a
signal is recovered.
Components of a PCM decoder
Time-Division Multiplexing
A time-division multiplex (TDM) system enables the joint
utilization of a common communication channel by a plurality of
independent message sources without mutual interference among them.
Each input message signal is first restricted in bandwidth by an
anti-aliasing LPF to remove the frequencies that are nonessential to
an adequate signal representation.

The LPF outputs are then applied to a commutator, which is
usually implemented using electronic switching circuitry.

The function of the commutator is twofold:

(1) To take a narrow sample of each of the N input messages at a rate f
s

that is slightly > 2W, where W is the cutoff frequency of the
anti-aliasing filter

(2) To sequentially interleave these N samples inside the sampling
interval T
s
.
(a)
A
(b)
Trunk
group
B
B
C
C
A
A
B
C
A
B
C
MUX MUX
Multiplexing

Block diagram of TDM system.
Interleaving
Synchronous time-division multiplexing
(b) Combined signal fits into channel bandwidth
A
C
B
f
C
f
B
f
A
f
H
H
H
0
0
0
(a) Individual signals occupy H Hz
Frequency-division Multiplexing
FDM
FDM demultiplexing
Analog hierarchy
Digital Multiplexers

The multiplexing of digital signals is accomplished by
using a bit-by-bit interleaving procedure with a selector switch
that sequentially takes a bit from each incoming line and then
applies it to the high-speed common line.

A worldwide feature of the hierarchy is that it starts at 64
kb/s, which corresponds to the standard PCM representation of a
voice signal.
An incoming bit stream at this rate, is called a digital signal
zero (DS0).
Digital hierarchy
T-1 line for multiplexing telephone lines