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Introduction To Signal Processing

The document provides an introduction to signal processing, covering topics such as the classification of signals, the sampling theorem, and digital signal processing techniques. It explains the importance of digitization for noise reduction, cost efficiency, and reliable data transfer. Additionally, it discusses key concepts like aliasing, quantization, and various parameters related to audio and video digitization.

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Jeara Muyong
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0% found this document useful (0 votes)
119 views10 pages

Introduction To Signal Processing

The document provides an introduction to signal processing, covering topics such as the classification of signals, the sampling theorem, and digital signal processing techniques. It explains the importance of digitization for noise reduction, cost efficiency, and reliable data transfer. Additionally, it discusses key concepts like aliasing, quantization, and various parameters related to audio and video digitization.

Uploaded by

Jeara Muyong
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd

Introduc)on to Signal Processing

Why Digi/ze?
Part 1: Introduc)on to Signal Processing
1. Classifica/on and Characteris/cs of Signals 1. To solve noise issues
2. Sampling Theorem and Aliasing
2. To reduce hardware costs
3. Difference Equa/ons for FIR and IIR Filters
4. Convolu/on and Correla/on 3. To store and reuse informa/on easily
Classifica)on and Characteris)cs of Signals 4. To achieve faster and more reliable data
transfer
Electronic Signal
• a /me varying voltage, current or EM wave that 5. To reduce system’s weight and size
poten/ally provides informa/on on the status of
a physical system, or conveys a message between
observers, among other possibili/es
Digi)za)on

• taking analog informa/on and encoding it into


1s and 0s so that computers can store, process,
and transmit such informa/on

✓ Sampling – process of measuring the


instantaneous values (samples) of con/nuous-
/me signal in a discrete form

✓ Quan/zing – process of discre/zing the sample


points of the sampled signal with a number of
quan/za/on levels

Analog signals – con/nuous-wave signals that


change with /me period and thus represented
with the sine waves

Digital signals – discrete in nature, represented


with square waves.
Common Basic Electronic Signals Spectrum
• used to classify something, or suggest that it
can be classified, in terms of its posi/on on a
scale between two extreme or opposite points
o Signal Spectrum – describes a signal's
magnitude and phase characteris/cs as a
func/on of frequency
o System Spectrum – describes how the
system changes signal magnitude and
phase as a func/on of frequency

The Audio Spectrum

Other Classifica/on of Signals

The Electromagne/c Spectrum

System
• may be referred to any set of
components, which func/on in
interrelated manner for a common cause
or objec/ve.
• has the following characteris/cs:
✓ Organiza/on
✓ Interac/on
✓ Interdependence
✓ Integra/on
✓ Central Objec/ve
Proper/es of a System Signal analysis is the study of collec/ng,
• Memory understanding, and deducing informa/on and
• Casualty intelligence from various signals
• Stability
• Inver/bility Signal processing covers the analysis,
• Time-Invariance modifica/on, and synthesis of signals.
• Linearity

Time Response
• occurs when the output of a system for an
input varies with respect to /me
• consists of two parts:
- Transient Response
- Stead-State Response

Basic Opera/ons of Signal Processing

Frequency Response
• quan/ta/ve measure of the output
spectrum of a system, that provides useful
insights into its stability and performance
characteris/cs
• system response to sinusoidal inputs at
varying frequencies
Digital Signal Processing (DSP) is the
mathema/cs, the algorithms, and the techniques
used to manipulate signals a`er they have been
converted into a digital form. This includes variety
of goals, such as:
1. Enhancement of Visual Images
2. Data Compression for Storage and
Transmission
3. Speech Recogni/on & Genera/on

Why DSP?
- highly linear (within limited dynamic
range), allowing noise to be control easily
a`er ini/al quan/za/on
- complex algorithms fit into a single chip,
while allowing flexibility for easy
adjustment of parameters via so`ware
- insensi/ve to component tolerances,
aging, environmental condi/ons,
electromagne/c interference
Sampling Theorem and Aliasing EXAMPLE:

Sampling Theorem A digital communica/on system is designed to


• specifies the minimum-sampling rate at ensure accurate representa/on of the audio
which a con/nuous-/me signal needs to signal to be transmiked with a maximum
be uniformly sampled so that the original frequency of 4 kHz. Calculate the minimum
signal can be completely recovered or sampling rate and the required bandwidth for the
reconstructed by these samples alone system to accurately capture and reproduce the
• states that a signal has to be sampled at audio signal without aliasing.
least with twice the frequency of the
original signal

𝑓S > 𝑓N = 2𝐵 or 𝑓N ≥ 2𝑓MAX and 𝑓S ≥ 2𝐵

• also known as the Shannon-Nyquist


Theorem (or Nyquist Theorem)
Informa=on Theory studies the quan/fica/on,
Sampling converts the independent variable storage, and communica/on of informa/on. It is
(/me) from con/nuous to discrete the mathema/cal treatment of the concepts,
parameters and rules governing the transmission
of messages through communica/on systems
How the Sample/Hold Process Works

Sampling Period
• the fixed gap between the samples of the
discre/zed signal
• reciprocal of the Nyquist Rate (𝑓N) or the
sampling rate at which the data in the
message signal should neither be lost nor
it should get over-lapped as described by
the Sampling Theorem
EXAMPLE: A wireless communica/on system is
designed to transmit digital data over a channel
Sampling Rate and Bandwidth in Wireless
with a signal-to-noise ra/o (SNR) of 20 dB. The
Communica/ons
system uses a sampling rate of 10,000 samples
per second. Calculate the maximum bandwidth of
the channel according to the Shannon-Hartley
theorem.
Aliasing An/-Aliasing Filter Design : Filter Parameters

• phenomenon of a 1. Passband
high-frequency 2. Stopband
component in the 3. Akenua/on
spectrum of a 4. Inser/on loss
signal, taking on 5. Impedance
the iden/ty of a 6. Ripple
low-frequency 7. Shape Factor (or Bandwidth Ra/o)
component in the 8. Quality Factor (Q)
spectrum of its 9. Poles and Zeros
sampled version 10. Envelope/Time Delay
• correc/ve measures to reduce its effect 11. Roll-off Rate (or Akenua/on Rate)
include: 12. Cutoff Frequency (-3dB Frequency)
o sampling the filtered signal at a rate 13. Center Frequency (f0)
slightly higher than the Nyquist rate 14. Bandwidth (BW)
o use of an an=-aliasing filter at the
transmiker side to eliminate
unwanted high frequency components

An/-Aliasing Filter Design: Types of Filter

Based on Frequency Response

EXAMPLE:
A digital data acquisi/on system is designed to
Based on Order sample an analog signal at a rate of 10,000
samples per second. The analog signal contains
frequency components up to 4 kHz. To avoid
aliasing during the sampling process, design an
an/-aliasing filter for the sampling process.

Based on Topology
Pulse Code Modula=on (PCM) is a standardized Quan)za)on Parameters
method used in telephone network to change an
analog signal to a digital one. Bit Depth
à The analog signal is first sampled at • no. of bits of informa/on per sample
sample rate of 8 kHz, then each sample is • directly corresponds to the resolu/on of
quan/zed into 1 of 256 levels and then each sample
encoded into digital 8-bit words.
Representa)on Levels
• discrete amplitudes of quan/zed output
• also known as reconstruc6on levels

Resolu)on
• number of representa6on levels over the
range of analog values

Quantum
• spacing between two adjacent
representa/on levels
• also known as step-size

Quan)za)on Error
• difference between an input value and its
quan/zed value

Dynamic Range
• ra/o of the highest amplitude to the
lowest amplitude of the quan/zed signal

Quan)za)on and Encoding


EXAMPLE: Standard Audio Digi)za)on Parameters

Calculate the number of quan/za/on levels


required for PCM encoding to ensure an adequate
fidelity during signal reconstruc/on of an analog
signal that has a bandwidth of 4 kHz at a rate of
20,000 samples per second. Assuming the signal
voltage range to be -5V to +5V, solve also for the
resolu/on, quan/za/on error, and dynamic range
of the quan/za/on process.

EXAMPLE: Digital Image Parameters


• Resolu=on – number of pixels in an image
A PCM system is used to digi/ze an analog audio or the measure of width and height of the
signal having a voltage range of -2V to +2V. image in terms of pixels
Determine the system's resolu/on at 8-bit • Pixels (Picture Elements) – likle dots that
quan/za/on for PCM encoding and write the make up the images on video frames
coded value of the given por/on of the signal • Color Depth (Bit Depth) – number of bits
used to indicate the color of a single pixel
• Aspect Ra=o – ra/o of image width to its
height, denoted by a colon in between
• Frame Rate (frames per second or FPS) is
the rate at which consecu/ve images
(frames) appear on a display
Resolu/on and Pixels

Screen Size

Aspect Ra/o

Frame Rate
• number of frames per second (FPS)
• frequency (rate) at which consecu/ve
images (frames) are captured or displayed.
Standard Video Digi/za/on Parameters

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