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DSP
DSP
INDUSTRIAL Y CONTROL
AUTOMTICO
PROCESAMIENTO DIGITAL
DE SEALES
Tomado de:
Digital Signal Processing
Engr. Abdul Rauf Khan Rajput
Parte 1
Definiciones
Aplicaciones
Signal:
A signal is defined as a function of one or more variables
which conveys information on the nature of a physical
phenomenon. The value of the function can be a real
valued scalar quantity, a complex valued quantity, or
perhaps a vector.
System:
A system is defined as an entity that manipulates one or
more signals to accomplish a function, thereby yielding
new signals.
Continuos-Time Signal:
A signal x(t) is said to be a continuous time signal if it is
defined for all time t.
Discrete-Time Signal:
A discrete time signal x[nT] has values specified only at
discrete points in time.
Signal Processing:
A system characterized by the type of operation that it
performs on the signal. For example, if the operation is
linear, the system is called linear. If the operation is nonlinear, the system is said to be non-linear, and so forth.
Such operations are usually referred to as Signal
Processing.
5
Analog output
signal
Analog
Signal Processor
Analog Signal Processing
Analog
input
signal
A/D
converter
Digital
Signal Processor
Digital Signal Processing
D/A
converter
Analog
output
signal
DSP Applications
Space
Medical
Telephone
Radar
Sonar
Ordnance Guidance
Secure communication
Scientific
Classification of Signals
Deterministic Signals
A deterministic signal behaves in a fixed known way with
respect to time. Thus, it can be modeled by a known
function of time t for continuous time signals, or a known
function of a sampler number n, and sampling spacing T
for discrete time signals.
even
odd
odd
11
Periodic Signals
A continuous signal x(t) is periodic if and only if there
exists a T > 0 such that
x(t + T) = x(t)
where T is the period of the signal in units of time.
f = 1/T is the frequency of the signal in Hz. W = 2/T is the
angular frequency in radians per second.
The discrete time signal x[nT] is periodic if and only if
there exists an N > 0 such that
x[nT + N] = x[nT]
where N is the period of the signal in number of sample
spacings.
Example:
Frequency = 5 Hz or 10 rad/s
0
0.2
0.4
12
T=1/f
13
1
0
-1
10
14
Parte 2
Operaciones entre Seales
15
17
0.6
0.5
0.4
exp(-2t)
exp(-0.5t)
0.3
0.2
0.1
0
0
10
15 18
x[n]
10
5
0
-3
10
-1
0
-1.5
5
-1
-0.5
0.5
1.5
0
-6
-4
-2
0
n
x[2n]
x[0.5n]
-2
19
Time Reversal
-5
0
x[-n]
0
0
0
2
n
20
Time Shift
A signal may be shifted in time by replacing the independent variable n by n-k,
where k is an integer. If k is a positive integer, the time shift results in a delay of
the signal by k units of time. If k is a negative integer, the time shift results in an
advance of the signal by |k| units in time.
x[n+3]
10
x[n-3]
x[n]
1
0.5
0 -2
1
0.5
0 -2
1
0.5
0 -2
10
n4
10
21
2. Addition:
Let x1 [n] and x2[n] denote a pair of discrete time signals. The signal y[n] obtained by the addition of
x1[n] + x2[n] is defined as
y[n] = x1[n] + x2[n]
3. Multiplication:
Let x1[n] and x2[n] denote a pair of discrete-time signals.
The signal y[n] resulting from the multiplication of the
x1[n] and x2[n] is defined by
y[n] = x1[n].x2[n]
Parte 3
Conversin
DSP
23
3. Coding:
x(t)
0101...
Sampler
Quantize r
Coder
A/D Converter
25
What is DSP?
Converting a continuously changing waveform
(analog) into a series of discrete levels (digital)
26
What is DSP?
The analog waveform is sliced into equal
segments and the waveform amplitude is
measured in the middle of each segment
The collection of measurements make up
the digital representation of the waveform
27
-1
-1.5
17
15
13
11
0.5
0.22
0.44
0.64
0.82
0.98
1.11
1.2
1.24
1.27
1.24
1.2
1.11
0.98
0.82
0.64
0.44
0.22
1.5
19
-0.22
-0.44 21
-0.64
-0.82
23
-0.98
-1.11
25
-1.2
-1.26
27
-1.28
-1.26
29
-1.2
-1.11
31
-0.98
-0.82
33
-0.64
-0.44 35
-0.22
37
-0.5
0
0
1
What is DSP?
-2
28
29
SW-8
V-high
SW-7
V-7
SW-6
V-6
SW-5
Output
V-5
SW-4
V-4
SW-3
V-3
SW-2
V-2
SW-1
V-1
V-low
30
Comparator
Output
Higher
Equal
Lower
31
Binary Search
Initial conditions
Analog
5-volts
3.42-volts
2.5-volts
Digital
256
Unknown
(175)
128
Voltage to be converted
3.42-volts
Equates to 175 binary
0-volts
34
Binary Search
Binary search algorithm:
High Low
Low NewGuess
2
First Guess:
Analog
5-volts
Digital
256
3.42-volts
unknown
128
256 0
0 128
2
Guess is Low
0-volts
0
35
Binary Search
New Guess (7):
Analog
5-volts
3.42-volts
176 172
172 174
2
Digital
256
unknown
174
Guess is Low
(but getting really,
really, close)
0-volts
0
36
Binary Search
New Guess (8):
176 174
174 175
2
Analog
5-volts
3.42-volts
Guess is Right On
0-volts
Digital
256
175!
0
37
Binary Search
The speed the binary search is
accomplished depends on:
The clock speed of the ADC
The number of bits resolution
Can be shortened by a good guess (but usually
is not worth the effort)
38
40
41
150
150
100
100
50
0
-50 0
10
20
30
40
Amplitude
Amplitude
Raw
50
0
-50 0
-100
-100
-150
-150
Time
10
20
30
40
Time
43
150
150
100
100
50
0
-50 0
10
20
30
40
Amplitude
Amplitude
Raw
50
0
-50 0
-100
-100
-150
-150
Time
10
20
30
40
Time
44
Sample Rate
High Bit
Count
Good
Duplication
Slow
Low Bit
Count
Poor
Duplication
Fast
High Sample
Rate
Good
Duplication
Slow
Low Sample
Rate
Poor
Duplication
Fast
45
46
analog signal
1
0.8
0.6
0.4
0.2
0
-0.2
-0.4
-0.6
-0.8
-1
0
1
0.8
0.6
0.4
0.2
0
-0.2
-0.4
-0.6
-0.8
-1
0
sampled signal
Uniform Sampling:
47
Uniform sampling
Uniform sampling is the most widely used sampling scheme.
This is described by the relation
x[n] = x[nT]
- <n<
where x(n) is the discrete time signal obtained by taking samples of the analogue signal x(t) every T seconds.
The time interval T between successive symbols is called the Sampling Period or Sampling interval and its reciprocal 1/T = Fs is called the Sampling Rate (samples per second) or the Sampling Frequency (Hertz).
A relationship between the time variables t and n of continuous time and discrete time signals respectively, can be obtained as
n
t nT
Fs
(1)
48
(2)
(3)
F
f
Fs
(4)
49
Fs
1
2
2T
(5)
or
Fs = 2 Fmax
(6)
Sampling Theorem:
If x(t) is bandlimited with no components of frequencies greater
than Fmax Hz, then it is completely specified by samples taken at
the uniform rate Fs > 2Fmax Hz.
The minimum sampling rate or minimum sampling frequency,
Fs = 2Fmax, is referred to as the Nyquist Rate or Nyquist
50
Frequency. The corresponding time interval is called the Nyquist
n cos
n
2
40
(ii)
50
n
cos
n
cos(
2
n
/
2
)
cos
n
2
2
40
x2 [ n] cos 2
As, Shows identical in [ x1(n) & x2(n)] sinusoidal signals & indistinguishable. Ambiguity
is there for samples values. x(t) yield same values as y(t) when two are sampled at Fs=40,
then
n0
n 0
-2
-1
52
2
1.8
1.6
1.4
1.2
1
0.8
0.6
0.4
0.2
00
n0
n0
7
53
6
5
4
3
2
1
00
54
-1<a<0
a>1
a<-1
55
a re
x
[
n
]
r
cos
n
R
as a function of n, and separately plotting the imaginary part
n
asx a[nfunction
] r n sinofnn. (see plots on the next slide)
I
56
xR[n] = (0.9)ncos(n/10)
0.5
0
-0.5
0
10
20
30
40
50
60
50
60
1
xI[n] = (0.9)nsin(n/10)
0.5
0
-0.5
0
10
20
30
40
57
|x[n]|
[n]
10
10
0
-
-
58
y[n]
59
H
H
a1
a2
y2[n]
63
64
65
z-transform
Transform techniques are an important role in the analysis of
signals and LTI system.
Z- transform plays the same role in the analysis of discrete time
signals and LTI system as Laplace transform does in the analysis of
continuous time signals and LTI system.
For example, we shall see that in the Z-domain (complex Z-plan)
the convolution of two time domain signals is equivalent to
multiplication of their corresponding Z-transform.
This property greatly simplifies the analysis of the response of LTI
system to various signals.
DSP
Slide 66
X (r e
x[n](r e
) n
That is, the z-transform is the Fourier transform of the sequence x[n]r - n . for r=1
this becomes the Fourier transform of x[n].
The Fourier transform therefore corresponds to the z-transform evaluated on the
unit circle:
DSP
Slide 67
z-transform(cont:
Slide 68
z-transform(cont:
(or converge) if
n
X ( z)
x[ n]r
x[ n] z
In specific cases the inner radius of this ring may include the origin, and the outer
radius may
, then
DSP extend
Slide 69 to infinity. If the ROC includes the unit circle
z 1
the Fourier transform will converge.
z-transform(cont:
Most useful z-transforms can be expressed in the form
X ( z)
P( z )
,
Q( z )
Slide 70
X ( z)
a u[n]z (az )
n
n 0
az 1
az 1 1.or
equivalently z a .
1
z
X ( z ) (az )
, z a,
1
n 0
1 az
za
1 n
Slide 71
(0 R z L ).
The Fourier transform of x[n] converges absolutely if and only if the ROC of
the z-transform includes the unit circle.
The ROC cannot contain any poles.
If x[n] is finite duration (ie. zero except on finite interval ( N1 n N 2 ).
at z=0 or
), then the ROC is the entire Z-plan except perhaps
z=
.
If x[n] is a right-sided sequence then the ROC extends outward from the
outermost finite pole to infinity.
If x[n] is left-sided then the ROC extends inward from the innermost nonzero
pole to z = 0.
A two-sided sequence (neither left nor right-sided) has a ROC consisting of a
ring in the z-plane, bounded on the interior and exterior by a pole (and not
containing any poles).
The
ROC
DSP
Slideis72a connected region.
X ( z)
1
1
1
z 1
2
, z
1
2,
1
a u[ n ]
,........ for z a .
1 az 1
By inspection we recognise that
n
1
x[n]
u[ n ],
2
n
Also, if X(z) is a sum of terms then one may be able to do a term-byterm DSP
inversion
Slide 73 by inspection, yielding x[n] as a sum of terms.
X ( z)
M
k 0
N
bk z
k
k
ak z
It is always possible to factorX(z) as
k 0
X(z)
b0
a0
1 c z
1 d z
M
k 1
N
k 1
Slide 74
The
inverse
z-transform
Partial fraction expansion (Continue:)
If M<N and the poles are all first order, then X(z) can be expressed
N
Ak
as
X(z)
,
1
k 1 1 d k z
in this case the coefficients A k are given by
A k 1 d k z 1 X ( z )
z dk
If M>N and the poles are first order, then an expression of the form
cab be used, and Brs be obtained by long division of the numerator.
M-N
X(z) Br z
Ak
1
1 dk z
The A k ' s can be obtained using M N
r 0
DSP
Slide 75
k 1
X(z)
M-N
B z
r 0
k 1, k i
Ak
1 dk z
Cm
m 1
1 d z
1 m
1 d k z 1
correspond to exponentia l sequences. For these terms the
ROC properties must be used to decide whether the sequences
are left - sided or right - sided.
DSP
Slide 76
C-
x[n] z
DSP
Slide 77
x[ n]
z X ( z ),
ROC
Rx
x1 [ n]
z X 1 ( z ),
x2 [ n]
z X 2 ( z ),
ROC
ROC
R x1
R x2
ROC contains R x1 R x1 .
ROC R x
Y ( z ) x[ n n ] z
0
DSP
n0
Slide 78
x[ m] z
n0
x[ m] z
n
X ( z ).
( m n0 )
C-
X [ z / z0 ],
z0
n
ROC z0 R x ,
where the notation z 0 Rx , indicates that the ROC is scaled by z (that is,
inner and outer radii of the ROC scale by z ). All pole-zero locations are
similarly scaled by a factor z0: if X(z) had a pole at z z then X(z/z0)
will have a pole at z=z0z1.
0
e x[n] X e
j 0 n
DSP
Slide 79
j ( 0 )
D-
Differentiation
dX ( z )
,
dz
nx[n]
z z
ROC R x .
X ( z)
x[ n ] z
n -
We have
dX ( z )
z
z (n) x[n]z n1 nx[n]z n z{nx[n]}.
n
dz
a u[ n]
z
to be
1
1 z 1
x[n] na n u[n]
z a,
d
1
az 1
X(z)
1
1 2
DSP Slide 80
dz 1 az
1 az
z a.
E-
Conjugation
This property is
x * [n] z X * ( z*),
FHere
ROC R x .
Time reversal.
x * [ n]
z X * (1 / z*),
ROC
1
.
Rx
The notation 1/Rx means that the ROC is inverted, so if Rx is the set
of values such that rR z rL , then the ROC is the set of values of z su
that 1 / r l z 1/rR .
1
a z
X ( z)
,
1 1
1 az 1 a z
1 1
DSP
Slide 81
z a Rx.
1
G-
Convolution
x1[n] * x2 [n]
z X 1 ( z ) X 2 ( z ),
Here
ROC contains
x * [ n]
z X * (1 / z*),
ROC
R x1 R x2 .
1
.
Rx
The z-transforms of the signal x1[n] =anu[n] and x2[n] = u[n] are
X 1 ( z)
a n z n
n 0
and
1
,
1
1 az
z a
1
,
z 1
1
n 0
1 az
For a 1, The z-transforms of the convolution y[n] = x 1[n] *x2[n] is
1
z2
Y ( z)
z 1
1
1
1 az 1 az z a z 1
.X 2 ( z)
1
Y DSP
( z ) Slide
82
1
1
1 az 1 az
z2
z a z 1
z 1
DSP
Slide 83
I-
s d j ,
we
have
( d j ) T
dT
j T
z e
e e
.
Therefore
z e dT and z T 2f/f s 2 / s ,
where ws is the sampling frequency. As varies from to , the s-plane is
mapped to the z-plane:
The j axis in the s-plane is mapped to the unit circle in the z-plane.
The left-hand s-plane is mapped to the inside of the unit circle.
The right-hand s-plane maps to the outside of the unit circle.
DSP
Slide 84