Lecture 6 - Analogy and Digital Systesm
Lecture 6 - Analogy and Digital Systesm
EE 8315
Wednesday, May 7, 2025
Mr. Cuthbert John Karawa
Lecture 6
Signal Conversion and Processing
College of Information and Communication Technology (CoICT)
Department Of Electronics And Telecommunications Engineering.
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Introduction
➢Digital signal processing is a technology that is widely
used in many applications, such as automotive, consumer,
graphics/imaging, industrial, instrumentation, medical,
military, telecommunications, and voice/speech
applications.
➢Digital signal processing incorporates mathematics,
software programming, and processing hardware to
manipulate analog signals.
Analog-to-Digital Conversion
➢In order to process signals using digital techniques, the
incoming analog signal must be converted into digital
form.
➢Sampling and Filtering.
✓An anti-aliasing filter and a sample-and-hold
circuit are two functions typically found in a digital
signal processing system.
✓The sample-and-hold function does two operations,
the first of which is sampling.
✓Sampling is the process of taking a sufficient number
of discrete values at points on a waveform that will
define the shape of the waveform.
Analog-to-Digital Conversion
➢Sampling converts an analog signal into a series of impulses,
each representing the amplitude of the signal at a given instant
in time.
Analog-to-Digital Conversion
➢There are certain criteria during sampled analogy that
must be met in order to accurately represent the original
signal.
➢All analog signals (except a pure sine wave) contain a
spectrum of component frequencies.
➢For a pure sine wave, these frequencies appear in
multiples called harmonics.
➢The harmonics of an analog signal are sine waves of
different frequencies and amplitudes.
Analog-to-Digital Conversion
➢When the harmonics of a given periodic waveform are
added, the result is the original signal.
➢Before a signal can be sampled, it must be passed
through a low-pass filter (anti-aliasing filter) to
eliminate harmonic frequencies above a certain value as
determined by the Nyquist frequency.
Analog-to-Digital Conversion
➢The sampling theorem states that, in order to represent
an analog signal, the sampling frequency, fsample, must be
at least twice the highest frequency component fa(max) of
the analog signal.
➢ The frequency fa(max) is known as the Nyquist
frequency.
𝒇𝒔𝒂𝒎𝒑𝒍𝒆 = 𝟐𝒇𝒂(𝒎𝒂𝒙)
➢Low-pass filtering is necessary to remove all frequency
components (harmonics) of the analog signal that exceed
the Nyquist frequency.
➢If there are any frequency components in the analog
signal that exceed the Nyquist frequency, an unwanted
condition known as aliasing will occur.
Analog-to-Digital Conversion
➢An alias is a signal produced when the sampling
frequency is not at least twice the signal frequency.
➢An alias signal has a frequency that is less than the
highest frequency in the analog signal being sampled and
therefore falls within the spectrum or frequency band of
the input analog signal causing distortion.
Quantization
➢The process of converting an analog value to a code is
called quantization.
➢Each sampled value of the analog signal to a binary
code.
➢The more bits that are used to represent a sampled value,
the more accurate is the representation.
Analog-to-Digital Conversion
Digital Signal Processing
➢Digital signal processing converts signals that naturally
occur in analog form, such as sound, video, and
information from sensors, to digital form and uses
digital techniques to enhance and modify analog signal
data for various applications.
➢It first translates a continuously varying analog signal
into a series of discrete levels.
Digital Signal Processing
➢Basic block diagram of a typical digital signal processing
system
Homework
➢Read Chapter 12 of Digital Fundamentals by Floyd
➢Methods of Analog-to-Digital Conversion.
➢Digital-to-Analog Conversion
➢Methods of Digital-to-Analog Conversion.
➢The Digital Signal Processor (DSP)
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