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Ch.1 Sampling and Reconstruction: X T Ae F X T Ae y T A T A e e X T X T

1) The document discusses sampling and reconstruction of signals. It covers topics like continuous-time sinusoidal signals, exponentially decaying sinusoids, the frequency response of linear systems, and the Fourier series and transform. 2) It describes the sampling theorem, which states that a signal must be sampled at least twice its highest frequency to perfectly reconstruct it. It also discusses concepts like aliasing and anti-aliasing filters. 3) The document analyzes ideal and staircase reconstructors for converting sampled signals back to continuous-time signals. It shows that the ideal reconstructor uses a sinc function as its impulse response, while the staircase uses a sinc function modulated by a complex exponential.

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43 views18 pages

Ch.1 Sampling and Reconstruction: X T Ae F X T Ae y T A T A e e X T X T

1) The document discusses sampling and reconstruction of signals. It covers topics like continuous-time sinusoidal signals, exponentially decaying sinusoids, the frequency response of linear systems, and the Fourier series and transform. 2) It describes the sampling theorem, which states that a signal must be sampled at least twice its highest frequency to perfectly reconstruct it. It also discusses concepts like aliasing and anti-aliasing filters. 3) The document analyzes ideal and staircase reconstructors for converting sampled signals back to continuous-time signals. It shows that the ideal reconstructor uses a sinc function as its impulse response, while the staircase uses a sinc function modulated by a complex exponential.

Uploaded by

youni1130
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as PDF, TXT or read online on Scribd
You are on page 1/ 18

CH.

1 SAMPLING AND RECONSTRUCTION


Reviews
1) Continuous-time Sinusoidal Signals

x( t)= Ae jt x* ( t)= Ae - jt

; = 2f

: radian frequency

jt - jt y( t)= A cos t = A } 2 {e + e * =1 2 {x( t)+ x ( t) }

2) Exponentially Decaying Sinusoid

x( t)= Ae ( - + j)t = Ae - te jt x* ( t)= Ae ( - - j)t = Ae - te - jt

; = 2f

: radian frequency

- t jt - jt y( t)= Ae - tcos t = 1 } 2 Ae {e + e * =1 2 {x( t)+ x ( t) }

3) Freqnency Response of Linear Systems

y( t) = h( t- )x( ) d = x(t- )h( ) d


-

Y ( s) = H( s)X ( s) x( t) = e jt,
-

s= j

Y ( ) = H( )X ( )

4) Steady-State Sinusoidal Response if

y( t) = e j( t - )h( ) d
= e jt = e jtH( )

h( )e - j d

- 1 -

By principle of superposition,

x( t) = A 1e j t + A 2e j t
1 2

y( t) = A 1e j tH( 1 )+ A 2e j tH( 1 )
1 2

6) Fourier Series

x( t) :

a periodic function with period T

x ( t) = c n e jn t
n =- T T

; 0=

where

1 cn = T

/2

- /2

x( t)e - j n tdt
0

7) Fourier Transform

x( t) :

X ( ) = x( t)e - jtdt
-

an energy signal

1 x( t) = 2 =

X ( )e jtd

X ( f )e j2ftdf

- 2 -

FT(Fourier Transform)
x(t )

~ x (t )

-T

-T / 2

T /2

x( t) :

an energy signal

Consider the Fourier Series representation:


x( t)=

c n e j nt n =-
0

(*)
-

where

1 cn = T 1 =T

T x( t)e j ntdt T x( t) e j ntdt


/2
0

- /2

() x( t) = x( t), if |t |T x( t) = 0, otherwise.
=>

From Eq. (*)


x( t)

X ( ) the envelope of Tc n = x( t)e - j tdt


-

1 X( n) cn = T 0

1 X ( n )e j nt = T 0 n =- = 21 X ( 0n )e j nt 0, n =-
0 0

1 = 0 ) (T 2

As T

, 0.
0

x( t) = 21

Therefore,

X ( )e jtd

///

Equivalently, FT can be represented in terms of the frequency:

1 x( t) = 2 = X ( ) =

X ( )e jtd

where

X ( f )e j2ftdf x( t)e - jtdt


- 3 -

Sampling Theorem
If the highest frequency contained in an analog signal is sampled at a rate sample values.

f s 2f max = 2B ,

then

x( t)

x( t)

is

f max = B

and the signal ///

can be exactly recovered from its

Example
speech audio video

f max =4 kHz f max =20 kHz f max =4 MHz

fs fs fs

8 kHz 40 kHz 8 MHz

The minimum sampling rate allowed by the sampling theorem: f s,min = 2f max ; Nyquist rate For arbitrary values of

f s/2 [ - f s/2, f s/2 ]

fs

, ; Nyquist frequency or folding frequency ; Nyquist interval

- 4 -

DSP Frequency Units


f
: ordinary frequency [ hertz = cycle/second ] : nomalized frequency [ cycle/sample ]
= 2f : radian frequency [ rad/second ]

f/f s

= 2f/f s : digital frequency in [ rad/sample ]

- 5 -

Spectra of Sampled Signals


x( t): sampled signal of x( t)=

x( t)

x( nT) ( t - n T ) -

DTFT(Discrete-Time Fourier Transform)


Def: the Fourier Transform of the sampled signal

X( f )

=
=

x( t)e - j2ftdt

x( nT) ( t - nT ) e - j 2ftdt - n =-

= x( nT)
n =- n =-

= x( nT)e - j 2fnT
Properties

( t - nT ) e - j 2ftdt

(*)

; DTFT

X ( f ) is computable from the knowledge of the sampled values. 1)


2) periodic in

with period of

f s:

X( f + m f s)

= x( nT)e - j2 ( f + mf s)nT
n =-

= x( nT)e - j2fnTe - j2f smnT


n =-

= x( nT)e - j2fnT n =- = X( f)
3) Inverse DTFT

;( ) e - j2mnf sT =1

X( f) Eq (*) may be thought of as a Fourier Series expansion of the periodic function


in the frequency domain. Thus,

x( nT ) = f1

f s f

s/2

- s /2

X ( f )e

j 2ff n s

df

4) Numerical Approximation

- 6 -

X ( f) = x( t)e - j2ftdt
-

x( nT)e - j 2fnTT = T X( f)
n =-

( f ) X ( f ) = lim TX T0
5) Approximation by keeping only a finite number of time samples

x( nT ) :

X( f)

X L ( f ) = x( nT)e - j 2fnT
n=0

L-1

; time windowing

6) Relationship with the Z-transform Z-transform of sequence

x( n ) : X ( z ) = x( n )z - n
n =-

X( f)

= x( nT)e - j 2fT n
n =-

= x( nT)e
n =-

f -j 2 f n
s

= x( nT)e - j n n =- = X ( z )| z = e j

- 7 -

Spectrum Replication
x( t): sampled signal of x( t)

x( t)

= x( nT) ( t - n T )
n =-

= x( t) ( t - nT ) n =- = x( t)s( t)
where

s( t) =

n =-

( t - nT )

Using Fourier series expansion, we may rewrite as

s( t) = c n e
n =- n =-

j 2 T nt

j 1 =T e

2 nt T

where

1 cn = T
Then,

T T

/2

- /2

( t) e

-j 2 T nt

1 dt = T
s

x( t)= x( t)s( t) =

X ( f )=

x( t)e j2nf t T n =-

X ( f - nf s) T n =-

- 8 -

Aliasing and Anti-aliasing Prefilter

- 9 -

Example 1.5.2

x( t): 1) The frequency spectrum of the sampled signal

Its magnitude is

2) The windowed spectrum:

= 0.2,f s = 1

Hz and 2 Hz,L = 10

- 10 -

3) Alternatively,

Combining the two expressions, we may obtain the following identity:

4) As

T0 ,

- 11 -

Sol)

(a) x=y=15dB (b) y = 15 +

A stop = x 50

- 12 -

Analog Reconstructors

Two reconstructors: - ideal reconstructor - staircase reconstructor

The sampled input:


y( t)=

y( nT) ( t - nT ) n =-

The reconstructed analog output:

ya ( t) = y( t)* h ( t) = y( )h ( t- )d
= = y( nT)
n =-

y( nT) ( t- nT ) h ( t- )d - n =-

= y( nT)h ( t - nT )
n =-

( t - nT ) h ( t - )d

Also note that

Y a ( f )= H( f ) Y( f )
Y ( f )=

Y ( f - nf s) T n =-

- 13 -

1) Ideal Reconstructor

H( f ) = T,
Then, for

{ 0,

|f |f s/2 o.w.

[- f s/2, f s/2] ,

1 Y( f) = Y( f) Y a ( f )= H( f ) Y ( f) = T T
The impulse response:

h ( t) = - H( f)e j2ftdf = - f /2 Te j2ftdf T ( e jf t - e - jf t) = 1 2j sin ( f t) = j2 s j2f st t sin ( f st) = = sinc( f st) f st


f s/2
s s s

- 14 -

2) The Staircase Reconstructor

h ( t) = u ( t)- u ( t - T ) H( f) = h( t)e - j 2ftdt = 1e - j 2ftdt

1 ( e - j 2fT -1) = 1 e - j fT ( e jfT - e - j 2fT ) = -j 2f j 2f 1 fT ) e - j fT = j 2 f e - j fT2j sin ( fT) = T sin (fT = T sinc( fT) e - j fT

- 15 -

Anti-image Postfilter:

Equalizer filter:
fT jfT H EQ ( f) = HT = ( f) sin ( fT ) e ,

for [- f s/2, f s/2]

- 16 -

- 17 -

Digital Equalization Filter and Anti-image Postfilter

A Typical DSP System

- 18 -

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