A reminder of sampling theory

FROM ANALOG TO DIGITAL

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 Signal digitization in your daily life
... without knowing it

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Analog-to-digital signal conversion
Conversion is characterized by two discretizations
– the 1st concerns time, and is called sampling: this consists in taking samples of the analog
signal at regularly spaced instants
– The 2nd concerns amplitude, and is called quantization: this consists in coding the signal
amplitude on a finite number of binary elements: CAN

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 Diagram of an analog signal digitization
chain

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 Choices to be made when digitizing an
analog signal
Digitizing accuracy through choice of quantization step
– This problem can easily be solved by increasing the number of bits in the ADC.

– The higher the number of bits, the greater the number of possible values, the greater the dynamic range and
the better the digitizing accuracy.

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 Choices to be made when digitizing an
analog signal
Discretization accuracy through choice of sampling frequency
– fe must be sufficiently high so as not to lose too much signalsignal information
–However, the higher fe (lower Te), the shorter the time available for digitalavailable for digital
processing, and the greater the number of samples to be processed
– the analog signal must be able to be reconstituted (interpolated) from the samples.
(there are an infinite number of signals passing through these samples).

How to choose the sampling frequency fe?

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 Assumption: s(t) signal is band-limited
Let s(t) be a continuous-time signal and S(f) its spectrum.S(f) is assumed to have bounded
support:
S(f)=0 pour |f|>fmax

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Sampling theorem (Shannon 1949)
A band-limited signal in the frequency interval [-fmax ; +fmax]can be reconstructed
(interpolated) exactly from its samples
if fe ≥2 fmax
The limiting frequency fe / 2 is called the Nyquist frequency.

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Claude Shannon
- American engineer and mathematician, born 1916
- Father of information theory, i.e. the mathematical
of information communication
- Inventor of the bit
- Was the 1st to understand that the world
could bedescribed with 0's and 1's

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Sampling - Notations

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Mathematical modeling of ideal sampling
The ideal sampling operation for an analog signal involves multiplying it by a Dirac comb

The ideal sampled signal se(t) is a continuous-time signal

- It consists of a train of equidistant Dirac pulses weighted at each instant t=kTe by the amplitude of s(kTe) at that
instant.
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 Graphic illustration of ideal sampling in
the time domain

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 Effect of ideal sampling in the frequency
domain
Let s(t) be a continuous-time signal and S(f) its spectrum.
- S(f) is assumed to have bounded support: S(f)=0 for |f|>fmax

The ideal sampled signal is written as :

𝑠𝑒 (𝑡) = 𝑠(𝑡) × 𝛿𝑇𝑒 (𝑡)

From the properties of the Fourier transform :

F 𝑠𝑒 (𝑡) = F 𝑠(𝑡) × 𝛿𝑇𝑒 (𝑡) = 𝑆(𝑓)∗ F 𝛿𝑇𝑒 (𝑡)

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 Effect of ideal sampling in the frequency
domain

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 Reminder - Graphic illustration of the convolution
of a signal by a Dirac pulse
Convolving a signal s(t) with a Dirac pulse delayed from t0 means shifting the signal from t0

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 Graphic illustration of ideal sampling in
the frequency domain

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 Spectral unbundling condition

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 Illustration of spectral folding

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 Ideal analog signal reproduction

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 Practical analog signal digitization chain
En pratique :
- It is essential to precede the sampling operation with a low-pass filter, known as an anti-
aliasing filter, with a cut-off frequency fc slightly lower than the Nyquist frequency fe / 2.

The practical chain for digitizing an analog signal is therefore made up of the following elements:

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 Spectacular effects of spectral folding
❖ Aircraft blades turning upside down
– www.youtube.com/watch?v=ByTsISFXUoY

❖Bicycle wheel turning upside down

– www.youtube.com/watch?v=bI8lrqBBAXQ
❖Hélicoptère en lévitation
– www.youtube.com/watch?v=yr3ngmRuGUc

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