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Chap2 Sampling

1) Signals must be converted from continuous-time to discrete-time through sampling in order to be processed by digital systems like microprocessors. 2) Sampling involves taking the amplitude of a signal at regular time intervals. The sampling rate is the number of samples taken per second. 3) According to the Shannon sampling theorem, a signal can be perfectly reconstructed from its samples if the sampling rate is at least twice the highest frequency component of the original signal. This is known as the Nyquist criterion.

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0% found this document useful (0 votes)
30 views

Chap2 Sampling

1) Signals must be converted from continuous-time to discrete-time through sampling in order to be processed by digital systems like microprocessors. 2) Sampling involves taking the amplitude of a signal at regular time intervals. The sampling rate is the number of samples taken per second. 3) According to the Shannon sampling theorem, a signal can be perfectly reconstructed from its samples if the sampling rate is at least twice the highest frequency component of the original signal. This is known as the Nyquist criterion.

Uploaded by

Youdas Tairi
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© © All Rights Reserved
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Download as PDF, TXT or read online on Scribd
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Chapter 2

Signal Sampling
and Quantization

CEN352 Dr. Nassim Ammour King Saud University 1


Introduction1
• Even most of signals are in continuous-time domain, they should be converted to a number at different discrete
time to be processed by a microprocessor.

• The process of converting these signals into digital form is called analog-to-digital (A/D) conversion.

• The reverse process of reconstructing an analog signal from its samples is known as digital-to-analog (D/A)
conversion.

maps the continuous


Sampler converts the takes the digital signal
amplitude into a discrete
continuous-time signal into a and produces a sequence
set of amplitudes
discrete-time sequence of binary code-words

Components of an analog-to-digital converter (ADC)

CEN352 Dr. Nassim Ammour King Saud University 2


Introduction2

8 Bit Code

Digital Processor

CEN352 Dr. Nassim Ammour King Saud University 3


Sampling
• Periodic or uniform sampling, a sequence of samples 𝑥[𝑛] is obtained from a continuous-time signal 𝑥𝑐 [𝑡] by
taking values at equally spaced points in time. T is the fixed time interval between samples, is known as the
the sampling period.

• The reciprocal 𝐹𝑠 is called sampling frequency (cycles per second or Hz) or sampling rate (samples per second).
Sampling rate Sampling period Example sampling period: T = 125 µs.
Sample per second (Hz) 𝐹𝑠 = 1 𝑇 (second)
sampling rate: 𝐹𝑠 =1/125µs = 8,000 samples per second (Hz).

Sample and Hold


CEN352 Dr. Nassim Ammour King Saud University 4
Sampling Process
input signal
• The sampling of a continuous-time signal 𝑥[𝑛]
is equivalent to multiply the signal with pulse
train signal 𝑆(𝑡). Pulse Train
signal

Sampled
signal

𝑥 𝑡 Input analog signal Sampled signal ∞

𝑥𝑠 𝑡 = 𝑥 𝑡 ∙ 𝑆(𝑡) = 𝑥𝑎 𝑛𝑇𝑠 𝛿(𝑡 − 𝑛𝑇𝑠 )


𝑆(𝑡) Pulse train 𝑛=−∞

CEN352 Dr. Nassim Ammour King Saud University 5


Sampling Process – frequency domain 1
Fourier Transform

• The signal 𝑥𝑐 [𝑡] and its spectrum 𝑋𝑐 (𝑗Ω) Inverse Fourier Transform
(1)

• The sequence 𝑥[𝑛] and its periodic spectrum 𝑋 (𝑗𝜔)


(2) Analog Frequency (Hz)

Normalized Frequency
𝑛 (1)(2) (Cycles/ Sample)
• Since 𝑥[𝑛] is related to 𝑥𝑐 𝑡 𝑤𝑖𝑡ℎ 𝑡 = 𝑛𝑇 =
𝐹𝑠
Sampling Frequency (Hz)

• The desired relationship between sampled signal spectrum 𝑋𝑠 𝐹 and the continuous signal spectrum 𝑋𝑐 (𝐹)

From spectral analysis , and after ∞ 𝑋𝑠 (𝐹): Sampled signal spectrum


1
some mathematical operations 𝑋𝑠 (𝐹) = 𝑋𝑐 𝐹 − 𝑘𝐹𝑠 𝑋𝑐 𝐹 : Original signal spectrum
𝑇
𝑘=−∞ 𝑋 𝐹 ± 𝑘𝐹𝑠 : Replica spectrum

CEN352 Dr. Nassim Ammour King Saud University 6


Sampling Process – frequency domain 2

1 1 1 1
𝑋𝑠 (𝐹) = 𝑋𝑐 𝐹 − 𝑘𝐹𝑠 𝑋𝑠 𝑓 = ⋯ + 𝑋𝑐 𝐹 + 𝐹𝑠 + 𝑋𝑐 𝐹 + 𝑋𝑐 𝐹 − 𝐹𝑠 + ⋯
𝑇 𝑇 𝑇 𝑇
𝑘=−∞

1
• Spectrum of 𝑥[𝑛] is obtained by scaling the spectrum of 𝑥𝑐 [𝑡], putting copies of the scaled spectrum 𝑋𝑐 (𝐹),
𝑇
1
at all integer multiples of the sampling frequency 𝐹𝑠 = 𝑇 .

• The spectrum of 𝑥[𝑛] can be readily sketched if 𝑥𝑐 (𝑡)is assumed to be band-limited. 𝑋𝑐 𝐹 = 0 𝑓𝑜𝑟 𝐹 > 𝐹𝐻

• Two conditions obviously are necessary to prevent


overlapping spectral bands:
1. The continuous-time signal must be band-limited.

2. The sampling frequency Ω𝑠 must be sufficiently large so that:

Spectrum of continuous-time band-limited signal 𝑥𝑐 (𝑡)

CEN352 Dr. Nassim Ammour King Saud University 7


Sampling Process – frequency domain 3
Case 1: Ω𝑠 > 2Ω𝐻

spectrum of discrete-time signal 𝑥 𝑛 = 𝑥𝑐 𝑛𝑇 with Ω𝑠 > 2Ω𝐻

• The sampling operation leaves the input spectrum 𝑋𝑐 (Ω) intact when Ω𝑠 > 2Ω𝐻 , therefore, it should be possible
to recover or reconstruct 𝑥𝑐 (𝑡) from the sequence 𝑥 𝑛 .

• Sampling at Ω𝑠 > 2Ω𝐻 creates a guard band which simplifies the reconstruction process in practical applications.

CEN352 Dr. Nassim Ammour King Saud University 8


Sampling Process – frequency domain 4
Case2: Ω𝑠 < 2Ω𝐻

spectrum of x[n], showing aliasing distortion, when s Ω𝑠 < 2Ω𝐻

• If Ω𝑠 < 2Ω𝐻 , the scaled copies of 𝑋𝑐 (Ω) overlap, so that when they are added together, 𝑋𝑐 (Ω) cannot be
recovered from 𝑋(Ω) .

• This effect, in which individual terms overlap is known as aliasing distortion or simply aliasing.

CEN352 Dr. Nassim Ammour King Saud University 9


Sampling Theorem 1
• Question: Are the samples 𝑥[𝑛] sufficient to describe uniquely the original continuous-time signal and, if so, how
can 𝑥𝑐 [𝑡] be reconstructed from 𝑥[𝑛] ? An infinite number of signals can generate the same set of samples.

• Answer: The response lies in the frequency domain, in the relation between the spectra of 𝑥𝑐 [𝑡] and 𝑥[𝑛] .

different continuous-time signals with the same set of sample values

CEN352 Dr. Nassim Ammour King Saud University 10


𝑇 = 0.01 sec → 𝐹𝑠 =
1
=
1
= 100 𝐻𝑧
Sampling Theorem 2
𝑇 0.01
One sample
each 0.01 s

The signal is under-sampled 2 fmax=180 > fs

CEN352 Dr. Nassim Ammour King Saud University 11


Sampling Theorem 3
• An analog signal can be perfectly recovered (reconstruction filter) as long as the sampling rate is at least twice
as large as the highest-frequency component of the analog signal to be sampled (Shannon sampling theorem).

• Let 𝑥𝑐 (𝑡) be a continuous-time band-limited signal with Fourier transform:

Then 𝑥𝑐 (𝑡) can be uniquely determined by its samples 𝑥 𝑛 = 𝑥𝑐 (𝑛𝑇), where 𝑛 = 0, ±1, ±2, … if the sampling
frequency Ω𝑠 satisfies the condition:

2𝜋
𝐹𝑠 = ≥ 2 𝐹𝑚𝑎𝑥
𝑇𝑠
𝐹𝑠
• Half of the sampling frequency is usually called the Nyquist frequency (Nyquist limit), or folding frequency.
2

Example: 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.

CEN352 Dr. Nassim Ammour King Saud University 12


Example 1
Problem:
Euler's
identity

Solution:
Using Euler’s identity,

Hence, the Fourier series coefficients are:

CEN352 Dr. Nassim Ammour King Saud University 13


Example 1 – Contd.

a.

b. After the analog signal is sampled at the rate of 8,000Hz, the


sampled signal spectrum and its replicas centered at the
frequencies ±n𝑓𝑠 , each with the scaled amplitude being 2.5/T .
Scaled
amplitude
of 2.5/T

Replicas, no
additional
information.
CEN352 Dr. Nassim Ammour King Saud University 14
Signal Reconstruction
(Digital-to-Analog Conversion)
• The reconstruction process (recovering the analog signal from its sampled signal) involves two steps.

• First: the samples 𝑥 𝑛 (digital signal )are converted


into a sequence of ideal impulses 𝑥𝑠 𝑡 , in which
each impulse has its amplitude proportional to digital
output 𝑥 𝑛 , and two consecutive impulses are
separated by a sampling period of T.

𝑥𝑠 𝑡 = 𝑥 𝑛 𝛿(𝑡 − 𝑛𝑇𝑠 )
𝑛=−∞

• Second: The analog reconstruction filter (ideal


low-pass filter) is applied to the ideally recovered
signal 𝑥𝑠 𝑡 to obtain the recovered analog signal.
The impulse response of the reconstruction filter
is 𝜋𝑡
sin( )
𝑇
ℎ𝑟 𝑡 = 𝜋𝑡 𝑠
𝑇𝑠 CEN352 Dr. Nassim Ammour King Saud University 15
Signal Reconstruction – Contd.
• Before applying the reconstruction filter,
a zero-order hold is used to interpolate
between the samples in xs(t).
Ideal low-pass
filter

• Reconstruction filter

An ideal low-pass reconstruction filter is


required to recover the analog signal
Spectrum ( an impractical case). Practical low-pass
filter

A practical low-pass reconstruction


(anti-image) filter can be designed to reject
all the images and achieve the original signal
spectrum. CEN352 Dr. Nassim Ammour King Saud University 16
Signal Reconstruction – Contd.

the condition of the Shannon sampling theorem


is violated. We can see the spectral overlapping
between the original baseband spectrum and the Aliasing
spectrum of the replica (add aliasing noise)

 Perfect reconstruction is not possible, even if we use ideal low pass filter.

 if an analog signal with a frequency f is under-sampled, the aliasing frequency


component 𝑓𝑎𝑙𝑖𝑎𝑠 in the baseband is simply given by:

CEN352 Dr. Nassim Ammour King Saud University 17


Example 2
Problem:

Solution:

Using the Euler’s identity:

CEN352 Dr. Nassim Ammour King Saud University 18


a.

b.
The Shannon sampling theory condition is satisfied

CEN352 Dr. Nassim Ammour King Saud University 19


Example 3
Problem:

Solution:
a. b.

CEN352 Dr. Nassim Ammour King Saud University 20


8 Quantization
• During the ADC process, amplitudes of the analog signal to be converted have infinite
precision.

• Quantization : The quantizer converts the continuous amplitude signal discrete amplitude
signal.

• Encoding: After quantization, each quantization level is assigned a unique binary code.

A block diagram for a DSP system

CEN352 Dr. Nassim Ammour King Saud University 21


8 Quantization – Contd.
• A unipolar quantizer deals with analog signals ranging from 0 volt to a positive reference voltage

∆ : Step size of quantizer (ADC resolution)


𝑥𝑚𝑎𝑥 − 𝑥𝑚𝑖𝑛
𝑥𝑚𝑎𝑥 : Max value of analog signal ∆=
𝐿
𝑥𝑚𝑖𝑛 : Min value of analog signal

𝐿: Number of quantization level


𝐿 = 2𝑚
𝑚 : Number of bits in ADC

𝑥 − 𝑥𝑚𝑖𝑛
𝑖 : Index corresponding to binary code 𝑖 = 𝑟𝑜𝑢𝑛𝑑

𝒙𝒒: Quantization level 𝑥𝑞 = 𝑥𝑚𝑖𝑛 + 𝑖 ∙ ∆ 𝑖 = 0,1, … , 𝐿 − 1

∆ ∆
𝑒𝑞 : Quantization error 𝑒𝑞 = 𝑥𝑞 − 𝑥 𝑤𝑖𝑡ℎ − ≤ 𝑒𝑞 ≤
2 2

CEN352 Dr. Nassim Ammour King Saud University 22


8 Quantization – Contd.
Example:3-bit ADC channel accepts analog input ranging from 0 to 5 volts,

• bipolar quantizer deals with analog signals ranging from a negative reference to a positive
reference.

CEN352 Dr. Nassim Ammour King Saud University 23


9 Periodicity
• In discrete-time case, a periodic sequence is a sequence for which
where the period N is necessarily an integer.

Discrete sinusoidal: 𝑥 𝑛 = 𝐴 𝑐𝑜𝑠(𝜔𝑛 + 𝜃)


Pulsation: 𝜔 (rad/sample)
Phase shift: 𝜃
𝜔
periodic sequence with N = 7 samples
Frequency: 𝑓 = cycle/sample
2𝜋
1
Period: 𝑁 = 𝑓

𝜋 𝜔 𝜋 1 1
Example 1: 𝜔= → 𝑓= = 4 = → 𝑁= =8
4 2𝜋 2𝜋 8 𝑓
Period of N = 8

CEN352 Dr. Nassim Ammour King Saud University 24


9 Periodicity – Contd.
Example 2: Not periodic with N = 8

3𝜋 𝜔 3𝜋 8 3 1 16
𝜔= → 𝑓= = = → 𝑁 = = 3×
8 2𝜋 2𝜋 16 𝑓 3

But periodic with N = 16 (Must be integer)

CEN352 Dr. Nassim Ammour King Saud University 25

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