DSP SYSTEM

• Digital Signal Processing System

• Analog to Digital Conversion
• Nyquist–Shannon Sampling Theorem
• Aliasing
• Sampling Effect in Time Domain
• Sampling Effect in Frequency Domain
• Anti Aliasing Filter
• Under-sampling
• Sampling of Band Limited Signals
• Over-sampling
• Digital to Analog Conversion
 Analog input Signal x(t) Analog output Signal y(t)
Analog
Signal Processor

Analog Signal Processing

Analog input Analog output

Signal x(t) Signal y(t)
A/D Digital D/A
converte Signal Processor converte
r r

Digital Signal Processing

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It consists of
• an analog filter called (anti-imaging) filter,
• an analog-to-digital conversion (ADC) unit,
• a digital signal (DS) processor,
• a digital-to-analog conversion (DAC) unit,
• and an analog filter called reconstruction (anti-image) filter.

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• Most signals of practical interest are analog in nature
Examples: Voice, Video, RADAR signals,Transducer/Sensor
output, Biological signals etc

• So in order to utilize those benefits, we need to convert our

analog signals into digital

• This process is called A/D conversion

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A/D conversion can be viewed as a three step process

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A/D conversion can be viewed as a three step process

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 Sampler

• Analog is continuous in time and continuous in

signal
amplitude.
• It means that it carries infinite information of time and infinite
information of amplitude.

• Analog (continuous-time) signal has some value defined

at every time instant, so it has infinite number of sample
points.

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 Sampler

• It is impossible to digitize an infinite number of points.

• The infinite points cannot be processed by the digital signal

(DS) processor or computer, since they require an infinite
amount of memory and infinite amount of processing power
for computations.

• Sampling is the process to reduce the time information or

sample points.

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 Sampler

• The first essential step in analog-to-digital (A/D) conversion is

to sample an analog signal.

• This step is performed by a sample and hold circuit, which

samples at regular intervals called sampling intervals.

• Sampling can take samples at a fixed time interval.

• The length of the sampling interval is the same as the

sampling period, and the reciprocal of the sampling period is
the sampling frequency fs.
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 Sampler

• After a brief acquisition time, during which a sample is

acquired, the sample and hold circuit holds the sample steady
for the remainder of the sampling interval.
• The hold time is needed to allow time for an A/D converter to
generate a digital code that best corresponds to the analog
sample.
• If x(t) is the input to the sampler, the output is x(nT), where T
is called the sampling interval or sampling period.
• After the sampling, the signal is called “discrete time
continuous signal” which is discrete in time and continuous
in amplitude.
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Sampler

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 Sampler
Figure below shows an analog (continuous-time) signal (solid
line) defined at every point over the time axis (horizontal line)
and amplitude axis (vertical line).
Hence, the analog signal contains an infinite number of points.

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• Each sample maintains its voltage level during the sampling
interval 𝑇 to give the ADC enough time to convert it.
• This process is called sample and hold.

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The sampling theorem guarantees that an analogue signal can be
perfectly recovered as long as the sampling rate is at least twice
as large as the highest-frequency component of the analogue
signal to be sampled.

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Examples

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Example: For the following analog signal, find the Nyquist sampling
rate, also determine the digital signal frequency and the digital
signal

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Example: Find the sampling frequency of the following signal.

So sampling frequency should be

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 Exercise

Determine the Nyquist sampling rate of a signal

x(t) = 3sin(5000t + 17o)

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 How many hertz can the human eye see?

• Most don't notice unless it is under 50 or 60 Hz.

• Generally, people notice when the frame-rate is less than the

refresh rate of the display.

• Depending on the type of CRT, you couldn't see flicker at 30

Hz or you could still see it at 120 Hz.

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• When the minimum sampling rate is not respected, distortion
called aliasing occurs.

• Aliasing causes high frequency signals to appear as

lower frequency signals.

• To be sure aliasing will not occur, sampling is always preceded

by low pass filtering.

• The low pass filter, called the anti-aliasing filter, removes all
frequencies above half the selected sampling rate.

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• Figure illustrates sampling a 40 Hz sinusoid
• The sampling interval between sample points is T = 0.01 second,
and the sampling rate is thus fs = 100 Hz.
• The sampling theorem condition is satisfied

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• Figure illustrates sampling a 90 Hz sinusoid
• The sampling interval between sample points is T = 0.01 second,
and the sampling rate is thus fs = 100 Hz.
• The sampling theorem condition is not satisfied

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Example of Aliasing in the time domain
of various sinusoidal signals ranging
from 10 kHz to 80 kHz with a sampling
frequency Fs = 40 kHz.

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• There are two complementary signal descriptions.
• Signals seen as projected onto time or frequency domains.

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• For analog signals, the frequency range is from -∞ Hz to ∞ Hz

• For digital signals, the frequency range is from 0 Hz to Fs/2 Hz

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• Sampling causes images of a signal’s spectrum to appear at
every multiple of the sampling frequency fs.

• For a signal with frequency f, the sampled spectrum

has frequency components at kfs ± f

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• A signal with no frequency component above a certain
maximum frequency is known as a band-limited signal.

• In our case we want to have a signal band-limited to ½ Fs.

• Some times higher frequency components (both harmonics

and noise) are added to the analog signal (practical signals are
not band-limited).

• In order to keep analog signal band-limited, we need a filter,

usually a low pass that stops all frequencies above ½ Fs.

• This is called an “Anti-Aliasing” filter. 35

• Anti-aliasing filters are analog filters.

• They process the signal before it is sampled.

• In most cases, they are also low-pass filters unless band-pass

sampling techniques are used.

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• If the sampling rate is lower than the required Nyquist rate, that
is fS < 2W, it is called under sampling.

• In under sampling images of high frequency signals erroneously

appear in the baseband (or Nyquist range) due to aliasing.

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Signals whose frequencies are restricted to a narrow band of
high frequencies can be sampled at a rate similar to twice the
Bandwidth (BW) instead of twice the maximum frequency.

Fs ≥ BW

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• Oversampling is defined as sampling above the minimum
Nyquist rate, that is, fS > 2fmax.

• Oversampling is useful because it creates space in the

spectrum that can reduce the demands on the analog anti-
aliasing filter.

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 Quantizer

• After the sampling, the discrete time continuous signal still

carry infinite information (can take any value) in terms of
amplitude.

• Quantization is the process to reduce infinite information of

the amplitude.

• Quantizer do the conversion of discrete time continuous

valued signal into a discrete-time discrete-value signal.

• The value of each signal sample is represented by a value

selected from a finite set of possible values.
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 Quantizer

• The A/D converter chooses a quantization level for

each analog sample.

• Number of levels of quantizer is equal to L = 2N

• An N-bit converter chooses among 2N possible quantization

levels.

• So 3 bitconverter has 8 quantization levels, and 4

bit converter has 8 quantization levels.
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The quantization step size or resolution is calculated as:
Δ = Q = R/2N
where
R is the full scale range of the analog signal (i.e. Ymax - Ymin)
N is the number of bits used by the converter

• Resolution of a quantizer is the distance between two

successive quantization levels
• More quantization levels, a better resolution!
• What's the downside of more quantization levels?

The strength of the signal compared to that of the quantization

errors is measured by dynamic range and signal-to-noise ratio.
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4-bit Quantizer

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4-bit Quantizer

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• The error caused by representing a continuous-valued signal
(infinite set) by a finite set of discrete-valued levels.

• The larger the number of quantization levels, the smaller the

quantization errors.

• The quantization error is calculated as the difference between

the quantized level and the true sample level.

• Most errors are limited in size to half a

quantization step Q or Δ .
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• Suppose a quantizer operation given by Q(.) is performed on
continuous-valued samples x[n] is given by Q(x[n]), then the
quantization error is given by

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• Lets consider the signal which is to be quantized.

In the figure, we can see that there is a difference between the

original signal (Blue Line) and the quantized signal (Red Lines).
This is the error produced while quantization
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Quantization error can be reduced, however, if the number of
quantization levels is increased as illustrated in the figure

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Block Diagram of D/A Conversion

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• Once signal processing is complete, digital-to-analog
digitalconversion must occur.
(D/A)
• This process begins by converting each digital code into an
analog voltage that is proportional in size to the number
represented by the code.
• This voltage is held steady through zero order hold until the next
code is available, one sampling interval later.
• This creates a staircase-like signal that contains frequencies
above W Hz.
• These signals are removed with a smoothing analog low pass
filter, the last step in D/A conversion.
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• In the frequency domain, the high frequency
present
elementsin the zero order hold signal appear as images, copies
of the original signal spectrum situated around integer
multiples of the sampling frequency.

• The smoothing analog filter removes these images and so is

given the name of Anti-Imaging Filter.

• Only the frequencies in the baseband, between 0 and fS/2 Hz,

remain.

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