Sampling and Pulse Modulation

EC-205
NIT SILCHAR
Analog-to-Digital Conversion
Noise Immunity:
Digital signals are more robust to noise and interference compared to
analog signals.
Efficient Storage & Transmission:
Digital data can be compressed and stored/transmitted efficiently.
Error Detection & Correction:
Easier implementation of error-checking and correction techniques.
Integration with Digital Systems:
Digital signals can be directly processed by microprocessors, DSPs, and
computers.
Security & Encryption:
Easier to implement secure and encrypted communication.
What is Sampling?
Definition:
Sampling is the process of converting a continuous-time analog signal
into a discrete-time signal by taking measurements at regular time
intervals.

Statement:
To reconstruct a signal without loss of information, the sampling rate
must be at least twice the maximum frequency present in the signal.
Sampling
Analog signal is sampled every Ts sec.
Ts is referred to as the sampling interval.
fs = 1/Ts is called the sampling rate or sampling frequency.
There are 3 sampling methods:
1. Ideal - an impulse at each sampling instant
2. Natural - a pulse of short width with varying amplitude
3. 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.
Sampling Methods
Differences
Feature Natural Sampling Flat-Top Sampling

The analog signal is sampled by multiplying it with a The sampled signal is held constant at each sampled value for
Definition
train of short pulses (non-ideal sampling). a short duration.

Sample follows the shape of the analog waveform

Output Shape Sample is a flat-top pulse at the instantaneous amplitude.
during the pulse width.

Realism More closely resembles the actual analog signal. Easier for ADCs to process due to uniform pulse levels.

Complexity Requires accurate pulse timing and analog switching. Easier to implement in practical circuits.

Introduces aperture effect (slight distortion), but easier for

Signal Distortion Less distortion, but harder to process digitally.
digitization.

Used In Theoretical and high-accuracy systems. Most real-world systems with sample-and-hold circuits.

4.6
 Nyquist Theorem
Note

According to the Nyquist theorem, the

sampling rate must be
at least 2 times the highest frequency
contained in the signal.
Nyquist Sampling Rate
Example
 For an intuitive example of the Nyquist theorem, let us sample a
simple sine wave at three sampling rates:
1. fs = 4f (2 times the Nyquist rate),
2. fs = 2f (Nyquist rate), and
3. One-half the Nyquist rate.

 Figure shows the sampling and the subsequent recovery of the signal.
 It can be seen that sampling at the Nyquist rate can create a good
approximation of the original sine wave (part a).
 Oversampling in part b can also create the same approximation, but it
is redundant and unnecessary.
 Sampling below the Nyquist rate (part c) does not produce a signal
that looks like the original sine wave.
Example
More Examples
A complex low-pass signal has a bandwidth of 200 kHz. What is the
minimum sampling rate for this signal?

Solution
The bandwidth of a low-pass signal is between 0 and f, where f is the
maximum frequency in the signal. Therefore, we can sample this signal
at 2 times the highest frequency (200 kHz). The sampling rate is
therefore 400,000 samples per second.
More Examples
A complex bandpass signal has a bandwidth of 200 kHz. What is the
minimum sampling rate for this signal?

Solution
We cannot find the minimum sampling rate in this case because we do
not know where the bandwidth starts or ends. We do not know the
maximum frequency in the signal.
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 
 = (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 amplitudes Vmin=-20V and
Vmax=+20V.
We want to use L=8 quantization levels.
Zone width = (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
Aliasing in Sampling
Aliasing:
When the sampling rate is too low, higher frequency components
appear as lower frequencies in the sampled signal.
Solution:
Use anti-aliasing filters before sampling
Follow Nyquist criterion
Assigning Codes
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:
nb = log2 L
Given our example, nb = 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.
Quantization and Encoding
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  which results in smaller errors.

BUT, the more zones the more bits required to encode the samples ->
higher bit rate
Bit rate and bandwidth
requirements of PCM
The bit rate of a PCM signal can be calculated from the number of bits
per sample x the sampling rate
Bit rate = nb x fs
The bandwidth required to transmit this signal depends on the type of
line encoding used. Refer to previous section for discussion and
formulas.
A digitized signal will always need more bandwidth than the original
analog signal. Price we pay for robustness and other features of digital
transmission.
Example
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:
Pulse Code Modulation
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, i.e. remove
high frequency components that affect the signal shape.
PCM Encoder
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 PCM Decoder
Delta Modulation
This scheme sends only the difference between pulses, if the pulse at
time tn+1 is higher in amplitude value than the pulse at time tn, then a
single bit, say a “1”, is used to indicate the positive value.
If the pulse is lower in value, resulting in a negative value, a “0” is
used.
This scheme works well for small changes in signal values between
samples.
If changes in amplitude are large, this will result in large errors.
Process of Delta Modulation
Delta Modulation
Delta Demodulation
Delta PCM (DPCM)
Instead of using one bit to indicate positive and negative differences,
we can use more bits -> quantization of the difference.

Each bit code is used to represent the value of the difference.

The more bits the more levels -> the higher the accuracy.
Introduction to Pulse Modulation
Definition:
Pulse modulation involves modulating an analog signal into a series of
pulses for transmission.
Why Pulse Modulation?
Efficient digital transmission
Combines benefits of analog input with digital channel compatibility
Foundation for modern digital communication systems
Types:
Analog Pulse Modulation: PAM, PWM, PPM
Digital Pulse Modulation: PCM, Delta Modulation (covered separately)

4.31
Pulse Amplitude Modulation (PAM)
Concept:
Amplitude of each pulse is proportional to the instantaneous value of
the analog signal.
Types:
Single Polarity PAM (pulses above zero only)
Double Polarity PAM (positive and negative pulses)
Applications:
Used in baseband transmission and as a precursor to digital modulation.

4.32
PAM Signal

4.33
Pulse Width Modulation (PWM)
Concept:
Width (duration) of each pulse varies with the instantaneous amplitude
of the analog signal. Pulse amplitude remains constant.
Advantages:
Better noise immunity than PAM
Efficient for power control (e.g., motor drivers)
Applications:
Used in power electronics, audio amplifiers, and control systems.

4.34
PWM Signal

4.35
Pulse Position Modulation (PPM)
Concept:
Position (timing) of each pulse is varied according to the analog signal’s
amplitude. Width and amplitude are constant.
Advantages:
Improved noise immunity
Requires synchronization at receiver
Applications:
Used in optical communication, IR remote control, satellite telemetry.

4.36
PPM Signal

4.37
Comparison
Feature PAM PWM PPM
Low – pulses are
High – depends Moderate – of fixed width;
on pulse width variable pulse spacing variation
Bandwidth Usage
and amplitude width increases makes it
variations spectral content bandwidth
efficient

Moderate – High – precise

Low – easy to
Signal Generation requires accurate timing control and
implement with
Complexity width control synchronization
sample-and-hold
circuitry needed

Better than PAM – Best – position-

Poor – amplitude
Noise amplitude based; less
sensitive, easily
Performance constant, better sensitive to
corrupted
tolerance amplitude noise

4.38