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ECE 308 -3

ECE 308
Sampling of Analog Signals
Quantization of Continuous-Amplitude
Signals

Z. Aliyazicioglu

Electrical and Computer Engineering Department


Cal Poly Pomona

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Sampling of Analog Signals


Example: xa (t ) = 3cos100 t
1. Find the minimum sampling rate required to avoid aliasing.
2. If Fs = 200 Hz, What is the discrete-time signal after sampling?
3. If Fs = 75Hz , What is the discrete-time signal after sampling?
4. What is the frequency F of a sinusoidal that yields sampling
identical to obtained in part c?

Solution:

a = 100 F = 50 Hz
The minimum sampling rate is Fs = 2 F = 100Hz

and the discrete-time signal is


100 1
x(n) = xa (nT ) = 3cos n = 3cos n = 3cos2 n
100 2
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Sampling of Analog Signals
Solution:
b If Fs = 200 Hz , the discrete-time signal is

100 1
x(n) = 3cos n = 3cos n = 3cos 2 n
200 2 4

c If Fs = 75 Hz , the discrete-time signal is

x( n ) = 3cos
100
n = 3cos
4
n = 3cos 2
2 n = 3cos 2 1 n
3
75 3 3
c For the sampling rate Fs = 75 Hz ,
1
F = fFs = f 75 and f = in part in (c). Hence
3
75 So, the analog sinusoidal signal is
F= = 25Hz
3
ya (t ) = 3cos 2 Ft
= 3cos50 t
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The Sampling Theorem

We must have some information about the analog signal


especially the frequency content of the signal, to select the
sampling period T or sampling rate Fs.

For example A speech signal goes below around 20Khz.


A TV signal is up to 5Mhz.
Any analog signal can be represented as sum of sinusoids of
different amplitudes, frequencies, and phases.
N
xa (t ) = Ai cos(2 Fi t + i )
i =1

where N the number of frequency components. Suppose that Nth


frequency do not exceed the largest frequency Fmax

Fi < Fmax

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2
The Sampling Theorem

To avoid the aliasing problem, is selected so that

Fs > 2 Fmax

The analog signal should be in the range of

1 F 1
fi = i
2 Fs 2

or in radians

i = 2 fi

The sampling rate FN = 2 Fmax is called the Nyquist rate.

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The Sampling Theorem


Example: Consider an analog signal
xa (t ) = 3cos50 t + 10sin 300 t + 3cos100 t

Solution

The frequencies in the analog signal


F1 = 25Hz F2 = 150Hz F3 = 50Hz

The largest frequency is


Fmax = F2 = 150Hz

The Nyquist rate is


FN = 2 Fmax = 300 Hz

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The Sampling Theorem
Example: The analog signal
xa (t ) = 3cos 2000 t + 5sin 6000 t 10cos12000 t

1. What is the Nyquist rate for this signal?


2. Using a sampling rate Fs = 5000 samples/s . What is the
discrete-time signal obtained after sampling?
3. What is the analog signal ya (t ) we can reconstruct from
the samples if we use ideal interpolation?
Solution
1. The frequencies of the analog signal are
F1 = 1 KHz F2 = 3 KHz F3 = 6KHz

The Nyquist rate is FN = 2 Fmax = 12 KHz

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The Sampling Theorem


2. For Fs = 5KHz n
x( n ) = xa (nT ) = xa
Fa
1 3 6
= 3cos 2 n + 5sin 2 n 10cos 2 n
5 5 5
1 2 1
= 3cos 2 n + 5sin 2 1 n 10cos 2 1 + n
5 5 5
1 2 1
= 3cos 2 n 5sin 2 n 10cos 2 n
5 5 5
1 2
= 7 cos 2 n 5sin 2 n
5 5
F
For Fs = 5KHz , the folding frequency is Fmax = s = 2.5KHz
2
Hence, F1 = 1 KHz is not effected by aliasing
F2 = 3 KHz is changed by the aliasing effect F2' = F2 Fs = 2 KHz

F3 = 6KHz is changed by the aliasing effect F3' = F3 Fs = 1 KHz

1 2 1
So that normalize frequencies are f1 = f2 = f3 =
5 5 5
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The Sampling Theorem

Solution (cont)

c. The analog signal that we can recover is


ya (t ) = 7cos 2000 t 5cos 4000 t

which is different than the original signal xa (t )

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Quantization of Continuous-Amplitude Signals

Converting a discrete-time continuous-amplitude signal into a digital


signal by expressing each sample value as a finite number of digits,
is called quantization.
The error between continuous-valued signal and a finite set of
discrete value levels signal is called quantization error or
quantization noise.
>> t=0:0.01:10;
The output of quantizer is xq (n) = Q [ x(n) ] >> x=0.9.^t;
>> plot (t,x)
>> hold on
The quantizer error is eq (n) = xq (n) x( n) >> n=0:10;
>> x=0.9.^n;
>> stem(t,x,'r')
Example:

Lets consider the discrete-time signal


as
0.9n n0
x ( n) =
0 n<0

The sampling frequency is Fs = 1Hz.


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Quantization of Continuous-Amplitude Signals

n xq (n) x ( n) eq (n)

0 1.0000 1.0000 0.0000


1 0.9000 0.9000 0.0000
2 0.8000 0.8100 -0.0100
3 0.7000 0.7290 -0.0290
4 0.7000 0.6561 0.0439
5 0.6000 0.5905 0.0095
>> t=0:0.01:10; 6 0.5000 0.5314 -0.0314
>> x=0.9.^t; 7 0.5000 0.4783 0.0217
>> plot (t,x)
>> hold on
8 0.4000 0.4305 -0.0305
>> t=0:10; 9 0.4000 0.3874 0.0126
>> x=0.9.^t; 10 0.3000 0.3487 -0.0487
>> y=0.1*round(10*x);
>> stem(t,y,'r')
>> grid on
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Quantization of Continuous-Amplitude Signals

Using rounding process for quantization. The other method is truncation ,


which discards the excess digits.

The values allowed in the digital signal are called quantization


level.
Distance between two quantization level is called quantization
step size or resolution
If we use rounding process the quantization error is the range of


eq (n)
2 2

If xmin and xmax represent the minimum and maximum value of x( n)


and L is number of quantization level, then

xmax xmin
=
L 1
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Quantization of Continuous-Amplitude Signals
In the example xmin = 0 , xmax = 1 , and, L = 11 , which leads to = 0.1 .

Note:

If L increases, decreases. Hence, the quantization error


eq (n) decreases and the accuracy of the quantizer increases.

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Quantization of Sinusoidal Signal

Lets look at the quantizer error by quantizing the analog sinusoidal


signal xa (t ).

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Quantization of Sinusoidal Signal
The analog signal xa (t ) is almost linear between quantization levels.
The quantization error

eq (t ) = xa (t ) xq (t )

eq(t)
/2
- t
-/2 0


Here eq (t ) = t t
2

The mean-square error power Pq is

Find discrete time signal x1(n) and x2(n)

1 t 3 2
2 2
1 1 1 2
Pq = e (t )dt =
2
eq2 (t )dt Pq = t dt = =
2
q
0 0
2 2 3 0 12

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Quantization of Sinusoidal Signal

For b bit the all range is 2A, then

2A
=
2b
Hence, the mean-square error power Pq for the signal xa (t ) is
4 A2 A2
Pq = 2b
=
(12)2 (3)22b

The average power of the signal xa (t ) is


2
1 Tp A
( A cos t ) dt = 2
2
Px =
Tp 0

The ratio of the signal average power to the noise power is the
signal-quantization noise ratio (SQNR) gives
Px 3 2b
SQNR = = 2 In dB, SQNR(dB) = 10log10 SQNR = 1.76 + 6.02b
Pq 2

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Digital-to-Analog Conversion
Some cases we may need to convert digital signal to analog
signal again.
The process of converting a digital signal into an analog signal is
called Digital-to-Analog (DAC).
All D/A converters use some kind of interpolation. A simple form
of D/A conversion is zero-order hold or staircase
approximation. Simply holds constant the value of one sample
until the next one is received.

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Digital-to-Analog Conversion
A Linear interpolation is connect successive samples with strait-
line. It needs T second delay so that has knowledge about next
sample values.

Better interpolation can be achieved by using more sophisticated


high-order interpolation techniques.
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Problem
Problem 1.7
An analog signal contains frequencies up to 10Khz.
a. What range of sampling frequencies allows exact
reconstruction of this signal from the samples?
b. Suppose that we sample this signal with a sampling
frequency Fs=8 KHz. Examine what happens to the
frequency F1=5Khz.
c. Repeat part (b) for a frequency F2=9Khz.

Solution 1.7 a Fmax = 10 Khz. Fs 2 Fmax = 20 Khz.

Fs = 8Khz. Fs
b Ffold = = 4 Khz.
2
So, F = 5 Khz will be alias of 3KHz

c F = 9 Khzwill be alias of 1KHz.

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Problem
Problem 1.15 xa (t ) = sin 2 F0t < t <
F0
and x ( n ) = x ( nT ) = sin 2 n
Fs
Fs = 5 Khz. and F0 = 0.5 Khz. 0 n 99
Solution 1.15

>> n=0:99;
>> x=sin(2*pi*0.1*n);
>> stem (n,x)

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