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Signal samples
x (t )
Sampling interval T
nT
0
x (t )
2T
4T
6T
8T
10T
12T
Analog signal
nT
0
2T
4T
6T
8T
10T
12T
3
Sampling theorem:
An analog signal can be in theory perfectly recovered as
long as the sampling rate is at least twice larger than the
highest frequency of the analog signal to be sampled.
Examples:
To sample a speech signal containing frequencies up to
4 kHz, the minimum sampling rate is chosen to be at
least 8 kHz, or 8,000 samples per second.
For an audio signal with frequencies up to 20 kHz,
sample the audio signal at the sampling rate of at least
40,000 samples per second, or 40 kHz.
4
Sampling condition is
satisfied.
fmax=40Hz, fs=100 Hz
40 Hz
-1
0
0.01
0.02
0.03
0.08
0.09
0.1
0.08
0.09
0.1
5
10 Hz
90 Hz
1
Voltage
Undersampling:
Signal aliasing would
occur when the
sampling condition is
not satisfied.
fmax=90 Hz, fs=100 Hz
-1
0
0.01
0.02
0.03
0.07
T
x (t )
ADC
encoding
x s ( t ) x ( t ) p( t )
x (t )
x s (0)
x s (T )
x s ( 2T )
t
Xs( f )
1
1
1
X ( f fs ) X ( f ) X ( f fs )
T
T
T
X( f )
1.0
a.
B f max
f
B
B
Lowpass Filter
1
T
b.
Over sampling
fs B
0
Xs( f )
Replica
(shifted version)
fs
2
fs
fs B
0
1
T
c.
Perfect sampling
fs B
fs
fs B
Folding frequency/Nyquist
limit
Xs( f )
f
fs B
fs
0
1
T
d.
Under sampling
fs
fs B
Xs( f )
f
fs B
f s B
fs B
fs B
fs
fs B
10
Signal Reconstruction
y s (t )
Digital signal
DAC
y ( n)
y ( n)
y(0)
y (t )
Lowpass
reconstruction
filter
y (t )
y s (t )
y s (0) y s (T )
y(1)
y(2)
y s ( 2T )
n
Digital signal
f s 2 f max
f max B
11 f
Signal Reconstruction
Signal Reconstruction
Signal Reconstruction
Sample and
hold
Digital value
ADC
coding
Xs( f )
X( f )
Xa
f
fc
fa fc fs
2
fs
fs fa
R1
Choose
14142
.
C2 2f c
b g
Vo
C2
R1 R2
C1
R2
C1
b g
R1 R2 C2 2f c
H( f ) ff f
Xa
s
a
% aliasing noise level
X( f ) ff
H( f ) ff
a
f
1 a
fc
2n
f fa
1 s
f
c
2n
16
Hold
Circuit
DAC
y ( n)
y s (t )
y ( n)
Equalizer
y H (t )
y H (t )
y s (t )
Antiimage
filter
y (t )
y (t )
sin( fT )
% distortion 1
fT
100%
ADC
Antialiasing
filter
Sample
and hold
Quantization
binary
encoder
Digital
signal
processor
DAC
Zerothorder
hold
Antiimage
filter
y (t )
111
110
101
100
011
010
001
000
/2
/ 2
xq
eq
x
x
19
2
3
4
5
6
7
Binary code
111
110
101
100
011
010
001
000
/2
/ 2
Quantization error:
3
4
eq
x
21
2
3
ADC
23
DAC
DAC conversion
Digital signal
Quantization
and coding
zeroth-order
hold
Antiimage
filter
Analog signal
Binary code
00001001
01001011
11010010
00001101
24
Quantization error:
eq xq x
Quantization bound:
Signal to noise power ratio:
SNR
eq
2
2
E x2
E eq2
1 N 1 2
x ( n)
N
SNR nN01
1
2
e
q ( n)
N n 0
N 1
2
x
( n)
n 0
N 1
2
e
q ( n)
n 0
xrms
25
SNRdB 10.79 20 log10
Quantized x(n)
2
1
0
-1
-2
-3
-4
-5
0.002
0.004
0.006
0.008
0.01 0.012
Time (sec.)
0.014
0.016
0.018
0.02
26
Quantized error
Quantized speech
Original speech
we.dat: we
5
-5
0.05
0.1
0.15
0.2
0.25
0.05
0.1
0.15
0.2
0.25
0.05
0.1
0.15
Time (sec.)
0.2
0.25
-5
1
-1
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