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Lecture 1 Introduction

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9 views39 pages

Lecture 1 Introduction

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Viet Nam National University Ho Chi Minh City

University of Science
Faculty of Electronics & Telecommunications

Lecture 1:

Introduction

Dang Le Khoa
Email: [email protected]
Outline
• Block Diagram of the DSP Systems
• Continuous time signals
➢ Mathematical representation of signals
➢ Some special signals

➢ Complex signals

➢ Sinusoidal signal

➢ Complex exponential signals

➢ Noise

• Sampling of Signals
➢ Sampling of continuous-time signals
➢ The sampling theorem
➢ Aliasing
➢ Reconstruction of Signals
Block Diagram of the DSP Systems

Faculty of Electronics & Telecommunications. HCMUS [3]


DSP Scenario
⚫ Modern systems generally…
– get a continuous-time signal form a sensor
– a continuous-time system modifies the signal
– an “analog-to-digital converter” (ADC) sample the signal to create
a discrete-time signal
– A discrete-time system to do the processing
– and then (if desired) convert back to analog

Faculty of Electronics & Telecommunications. HCMUS [4]


Why DSP?
⚫ Digital data storage and transmission is much more effective
than in the analogue form
⚫ Flexibility: processing functions can be altered or adjusted
⚫ Possibility of implementing much more complicated processing
functions than in analogue devices
⚫ Efficient implementation of fast algorithms and matrix-based
processing
⚫ Speed of digital operation tends to grow rapidly with the years
of technical progress
⚫ A very high accuracy and reliability is possible to achieve
dynamic range can be increased
⚫ Signal multiplexing: simultaneous (parallel) processing
Faculty of Electronics & Telecommunications. HCMUS [5]
Applications
⚫ Music: recording, playback, mixing, synthesis, storage (e.g. CD-
players)
⚫ Speech: recognition, synthesis (e.g. automatics peakers)
⚫ Communications and multimedia: signal generation,
transmission, modulation and compression, data protection via
error correcting signal coding, (e.g. digital modems, TV and
telephony, computers, video conferencing and lnternet)
⚫ Radar: filtering, detection, feature extraction, localization,
tracking, identification (e.g. air-traffic control)
⚫ Image processing: 2D filtering, enhancement, compression,
pattern recognition (e.g.satellite images)
⚫ Biomedicine: diagnosis, patient monitoring, preventive care

Faculty of Electronics & Telecommunications. HCMUS [6]


Definition

⚫ Signal: Signals represent information about data, voice, audio,


image, video… There are many ways to classify signals but here
we categorize signals as either analog (continuous-time) or
digital (discrete-time).
⚫ Processing: Signal processing is to use circuits and systems
(hardware and software) to act on input signal to give output
signal which differs from the input, the way we would like to.

Faculty of Electronics & Telecommunications. HCMUS [7]


Continuous time signals

Faculty of Electronics & Telecommunications. HCMUS [8]


Mathematical representation of signals
⚫ Instead of describing signals by words or by plotting their
waveforms, the more objective and concise way is to express
signals mathematically, whenever possible.
⚫ Mathematical representation of signals in time domain and
transform domain is needed for analysis and design of circuits
and systems.

Faculty of Electronics & Telecommunications. HCMUS [9]


Sinusoidal signal
⚫ Sinusoidal signals (sinusoids or sine waves) are the most
popular analog signal
x(t ) = A cos(t +  0 )
A is the peak value, Ω angular frequency (radians/s),
 = 2 F , F is frequency (Hz), T = 1 / F the period (sec)
 0 phase at t=0

A cos( 0 )

Faculty of Electronics & Telecommunications. HCMUS [10]


Square wave
⚫ For the symmetric square wave , the mathematical expression
consists of one part for amplitude, and the other for periodicity

 T
− A, − 2  t  0
x(t ) = 
+ A,0  t  T
 2

Faculty of Electronics & Telecommunications. HCMUS [11]


Unit impulse
⚫ The unit impulse (delta Dirac function) is evolved from a
symmetric rectangular pulse of width  and amplitude 1 / 
when  → 0
+
 , t = 0
 (t ) = 
0, t  0

−
 (t )d (t ) = 1

Faculty of Electronics & Telecommunications. HCMUS [12]


Unit step
⚫ The signal rises suddenly from 0 to 1 at time t = 0 then remains
unchanged, similarly to the closure of an electric switch. Its
mathematical definition is
0, t  0
u (t ) = 
1, t  0

Faculty of Electronics & Telecommunications. HCMUS [13]


Complex signals
⚫ Natural physical quantities, signals included, are real-valued.
However sometimes the imaginary operator j = −1 is appended
to them by reason of mathematical convenience, such as to take
into account the phase difference between voltages and currents
in AC circuits. Following is an example of a complex signal:
x(t ) = 5cos t − j 5sin t
⚫ A complex signal comprises a real and an imaginary part:
x(t ) = xR (t ) + jxI (t ) = x(t ) e j (t )
x(t ) = xR2 (t ) + xI2 (t )
xI (t )
 (t ) = tan −1
xR (t )
x  (t ) = xR (t ) − jxI (t ) = x(t ) e − j (t )

Faculty of Electronics & Telecommunications. HCMUS [14]


Complex exponential signals
⚫ Complex exponentials, also called complex sinusoids, are more
often used. The general expression is
x(t ) = Ae j ( t +0 )
⚫ Phasor is the vector representation of the signal. It is periodic
with an angular period of 2 radians.
⚫ Công thức Euler
e jx = cos( x) + j sin( x)

0

Faculty of Electronics & Telecommunications. HCMUS [15]


Noise
⚫ All the unwelcome signals of random nature superimposing on
our signal carrying information are named together as noise.
⚫ The most convenient to model, also the most frequently
mentioned, is white noise which has the power spectral density
S(F) constant with frequency F

Faculty of Electronics & Telecommunications. HCMUS [16]


Noise…
⚫ Gaussian distribution: In reality, many random variables have
Gaussian distribution (also called normal distribution). Gaussian
white noise PDF and CDF are, respectively,
1 − x 2 /2 2
p( x) = e
2


m = E[ x] =  xp( x)dx
−


 = E[( x − m ) ] =  ( x − m)
2 2 2
p ( x)dx
−

Faculty of Electronics & Telecommunications. HCMUS [17]


Sampling of Signals

Faculty of Electronics & Telecommunications. HCMUS [18]


Sampling of continuous-time signals
⚫ Sampling a continuous-time signal turns it into a corresponding
discrete-time signal so that it can be processed on digital
systems. Actually, the sampling is followed by two other
operations, quantization and binary encoding. In reality, the
analog-to-digital converters (abbreviated ADC or A/D) do all
the three steps.

The sampling of a signal at regular interval t = nT where n


is an integer, positive and negative, that is, n = 0, 1, 2,.., -1, -
2,… This is uniform sampling that we use routinely
Faculty of Electronics & Telecommunications. HCMUS [19]
Sampling of continuous-time signals

▪ The process converts the analog signal into a digital signal.

xa ( t ) x (n) xq ( n ) xd ( n ) = [0,1,...]
Sampling Quantization Encoding

▪ Sampling
Convert an
xa ( t ) analog signal xa ( nT ) = x ( n )
to sampling

sa ( t )

Faculty of Electronics & Telecommunications. HCMUS [20]


Sampling of continuous-time signals…
⚫ The time distance T is called sampling interval or sampling
period, Fs = 1 / T is sampling frequency (Hz or samples/sec) or
sampling rate.

Faculty of Electronics & Telecommunications. HCMUS [21]


Sampling of continuous-time signals…
⚫ Figure illustrates the process, and an electric switch as a way to
implement the sampling: when the contact closes in a short time,
the signal passes; and when the contact opens, no output signal
appear

⚫ The samples are written as x(t ) or x(nT ) but T usually taken


as 1, the samples wil be denoted as x( n) . The integer n is called
index.
Faculty of Electronics & Telecommunications. HCMUS [22]
Sampling of continuous-time signals…

xa ( t ) sa (t ) =   (t − nT )
n =−

t t
0 0 T 2T …
Analog Signal Sampling signal
▪ Analog Signal: xa ( t )

▪ Sampling signal: sa ( t ) =   ( t − nT )
n =−

t0 +
, t = 0, 
 (t ) =    ( t )dt = 1   s ( t )  ( t − t )dt = s ( t )
 0, t  0.
0 0
− t0 −

Faculty of Electronics & Telecommunications. HCMUS [23]


Sampling of continuous-time signals…
xs ( t ) xa ( nT ) = x ( n )

n n
0 T 2T … 0 T 2T …
The samples nth dircet time signal
▪ The samples:
 
xs ( t ) = xa ( t ) sa ( t ) =  x (t )  (t − nT ) =  x ( nT )  (t − nT )
n =−
a
n =−
a

▪ nth dircet time signal


nT +
x (n) =   x ( t ) dt = x ( nT )
s a
nT −

Faculty of Electronics & Telecommunications. HCMUS [24]


Relations Among Frequency Variables

Faculty of Electronics & Telecommunications. HCMUS [26]


Example
⚫ The implications of these frequency relations can be fully
appreciated by considering the two analog sinusoidal signals
x1 (t ) = cos 2 (10)t
x2 (t ) = cos 2 (50)t

which are sampled at a rate Fs = 40 Hz. The corresponding


discrete-time signals or sequences

10 
x1 (t ) = cos 2 ( )n = cos n
40 2
50 5
x2 (t ) = cos 2 ( )n = cos n
40 2
5
⚫ However, x2 (t ) = cos n = cos(2 n +  n / 2) = cos( n / 2) : Aliasing
2

Faculty of Electronics & Telecommunications. HCMUS [27]


The sampling theorem
⚫ Let’s consider a continuous-time signal x(t) representing certain
information such as voice. Its magnitude frequency spectrum
|X(F)| where FM is its maximum frequency. The signal is
sampled by a sequence of narrow pulses  (t ) of amplitude 1.
⚫ The spectrum bands do not overlap so we can recover the analog
signal by lowpass filtering the central band, or bandpass filtering
any other bands.

Faculty of Electronics & Telecommunications. HCMUS [28]


Faculty of Electronics & Telecommunications. HCMUS [29]
The sampling theorem
In order that the samples represent correctly the original
analog signal, the sampling frequency must be greater than
twice the maximum frequency component of the analog signal:

Fs  2 FM

The limiting frequency 2FM is called Nyquist rate, and the central
frequency interval [-Fs/2, Fs/2] is called the Nyquist interval.

Faculty of Electronics & Telecommunications. HCMUS [30]


Examples
⚫ A waveform contains the fundamental frequency of 1 kHz and a
second harmonic 2 kHz, then the sampling rate must be greater
than 2 x 2 kHz = 4 kHz, say 5 kHz or more.
⚫ Another example is the voice in the telephone system. The voice
is limited by a high quality analog filter at FM = 3.4 kHz, then
the sampling frequency must be greater than 2 x 3.4 = 6.8 kHz,
say 8 kHz or more.

Faculty of Electronics & Telecommunications. HCMUS [31]


Aliasing
⚫ The low frequency signal x1 (t ) is sampled 4 times at S1, S2, S3
and S4 in a period of the signal, that is, Fs = 4Fx1. From these
samples we would be able to recover x1 (t ) .
⚫ For the high frequency signal x2 (t ) there are the same 4 samples
S1, S2, S3 and S4 in its 9 cycles, so the sampling frequency is
just (4/9)Fx2 . From these sample points of x2(t) we will
recover x1(t) and not the original x2(t).
⚫ Thus the high frequency signal when undersampled will be
recovered as a low frequency signal. This phenomenon is called
aliasing, and the recovered low frequency, which is false, is
called the alias of the original high frequency signal.

Faculty of Electronics & Telecommunications. HCMUS [32]


Aliasing…
⚫ To avoid aliasing, there are two approaches: One is to raise the
sampling frequency to satisfy the sampling theorem; the other is
to filter off the unnecessary high-frequency content from the
continuous-time signal. We limit the signal frequency with an
effective lowpass filter, called an antialiasing prefilter, so that
the remaining highest frequency is less than half of the intended
sampling rate. If the filter is not perfect, we must give some
allowance.
⚫ For example in voice processing, if the lowpass filter still allows
frequencies above 3,4kHz go through even at small amplitudes,
the sampling frequency should be 8 kHz or higher.

Faculty of Electronics & Telecommunications. HCMUS [33]


Example
⚫ A signal at frequency Fx = 50Hz. What frequency will be
recovered? (a) Fs = 120 Hz, (b) Fs = 80 Hz
Solution

(a) With Fs = 120 Hz, The Nyquist interval is

𝐹𝑠 𝐹𝑠 120 120
− , = − , 𝐻𝑧 = −60, 60 𝐻𝑧
2 2 2 2

The sampling theorem is satisfied, then the original frequency of


50 Hz will be recovered.

Faculty of Electronics & Telecommunications. HCMUS [34]


Example…
(b) With fs = 80 Hz, The Nyquist interval is

𝐹𝑠 𝐹𝑠 80 80
− , = − , 𝐻𝑧 = −40,40 𝐻𝑧
2 2 2 2
Signal Fx = 50 Hz outside the Nyquist interval, we can not recover
this signal. The recovered signal will be frequencies
F0 = F ± mfs = 50 ± m80=50, 50 ± 80, 50 ± 160, 50 ± 240
Only the frequency -30 Hz lies within the Nyquist interval, then the
recovered signal will be -30 Hz (30Hz and phase reversal).

35
CHƯƠNG 1
Lowpass filter

Application
⚫ Anti-aliasing prefilter
⚫ Postfilter

Faculty of Electronics & Telecommunications. HCMUS [36]


Example
⚫ Consider the signal
x(t ) = 4 + 3cos  t + 2cos 2 t + cos3 t (t:ms)
(a) Find the Nyquist frequency.
(b) If the signal is sampled at half the Nyquist frequency, find the
signal x0(t) that is the alias of x(t)

Faculty of Electronics & Telecommunications. HCMUS [37]


Example…
⚫ Solution
(a) Because the unit of time is ms, the given signal has 4 frequencies
F1 = 0Hz, F2 = 0.5kHz, F3 = 1kHz, F4 = 1.5kHz
The highest frequency is FM = F4 = 1.5kHz, the Nyquist rate is 2x1.5kHz = 3kHz. When
the signal is sampled at rates greater than 3kHz there will be no aliasing.
(b) When the signal is sampled at 1.5kHz aliasing will occur. Now the Nyquist interval is
(-0.75, 0.75)kHz. The two frequencies F1 and F2 lie within this interval and, thus, will
not be aliased. The two frequencies F3 and F4 lie outside the Nyquist interval and, thus,
will be aliased:
F30 = F3  mFs = 1mod(1.5) = 1 – 1.5 = - 0.5kHz
F40 = F4  mFs = 1.5mod(1.5) = 1.5 – 1.5 = 0kHz
The recovered signal x0(t) has the frequencies F10, F20, F30 and F40
x0(t) = 4cos2F1t + 3cos2F2t + 2cos2F30t + cos2F40t
= 4 + 3cost + 2cos(-t) + cos0 = 5 + 5cost

Faculty of Electronics & Telecommunications. HCMUS [38]


Reconstruction of Signals
⚫ An oversampling D/A converter are subdivided into a digital
front end followed by an analog section. The digital section
consists of an interpolator whose function is to increase the
sampling rate by some factor I. The interpolator simply
increases the digital sampling rate by inserting I −1 zeros
between successive low rate samples.
⚫ The resulting signal is then processed by a digital filter with
cutoff frequency Fc = B/Fs in order to reject the images
(replicas) of the input signal spectrum. The output analog filters
have a passband of 0 ≤ F ≤ B hertz and serve to smooth the
signal and to remove the quantization noise in the frequency
band B ≤ F ≤ Fs/2.

Faculty of Electronics & Telecommunications. HCMUS [39]


Faculty of Electronics & Telecommunications. HCMUS [40]

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