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Sampling - (Quantization - Reconstruction)

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15 views13 pages

Sampling - (Quantization - Reconstruction)

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ramymohamed7801
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Subject: Communications 2

Code: EEC 223

Presented By:

Dr. Mohamed Youssef

E-mail:
[email protected]
Agenda

• Quantization (Cont.).

• Uniform Quantization.

• Signal Reconstruction: “Interpolation Formula”

• Practical Difficulties in Signal Reconstruction:

• The Treachery of Aliasing

• A Solution: Antialiasing Filter


Quantization (Cont.)

❑ Quantization principle is based on that any human sense (ear & eye) can only detect finite intensity differences.

❑ There are two types of quantization:

➢ Uniform quantization: The representation levels are equally spaced (Uniformly spaced) ….
Used for biomedicine, audio systems.

o Midrise
o Midtread

➢ Non-uniform quantization: The representation levels have variable spacing from one another
Used for communication systems for the need to compress signals.
❑ The full-scale range of the I/P signal is divided into 2n values; (n is the number of ADC bits).
Uniform ❑ Each analog sample is assigned to one of the 2n values by truncation.
Quantization
❑ The difference between two adjacent values is called ‘Quantum’ or step size (a).
Signal Reconstruction: “Interpolation Formula”

❑ The Process of reconstructing a continuous-time signal g(t) from its samples is also knows as interpolation.

❑ We saw that a signal g(t) band-limited to B Hz can be reconstructed (interpolated) exactly from its samples.

❑ This is done by passing the sampled signal through as ideal low-pass filter of bandwidth B Hz.

❑ As seen from the following equation, the sampled signal must be passed through as ideal low-pass filter of
bandwidth B Hz and gain Ts.

➢ Thus, the reconstruction (or interpolating) filter transfer function is


Signal
Reconstruction 𝒇
(Cont.): 𝑨 𝒓𝒆𝒄𝒕
𝑻
=⇒ 𝑨𝑻 𝐬𝐢𝐧𝐜 𝐭𝐓П

The impulse response of this filter , the inverse Fourier transform of H(f):

Assuming the use of Nyquist sampling rate fs = 2B ➔ 2BTs = 1 ➔ then,



𝑔𝑠 𝑡 = ෍ 𝑔 𝑛𝑇𝑠 δ 𝑡 − 𝑛𝑇𝑠
𝑛=−∞
Find a signal g(t) that is band-limited to B Hz and whose samples are g(0) = 1 and
g(±Ts) = g(±2Ts) = g(±3Ts) = …. 0 where the sampling interval Ts is the Nyquist
interval, that is Ts = 1/2B.

Example: ➢ we use the interpolation formula to construct g(t) from its samples.

➢ Note: only signal that has a bandwidth B Hz and with the sample values g(0) = 1.
➢ g(nTs ) = 0 (n ≠ 0)
Practical Difficulties in Signal Reconstruction:
❑ If a signal is sampled at the Nyquist rate fs = 2B Hz, the spectrum of
sampled signal consists of repetitions of the original one without any gap
between successive cycles as shown in the following figure.
❑ To recover the original signal, we need to pass the sampled signal through an ideal low-pass filter shown
dotted in the previous figure. But this filter is unrealizable (it can be closely approximated only with infinite
time delay in the response).
❑ A practical solution to this problem is to sample the
signal at a rate higher than the Nyquist rate (fs > 2B Hz).
❑ We can now recover the original signal using low-pass
filter with gradual cutoff characteristic shown dotted in
the following figure.
❑ But even in this case, the filter gain is required to be zero beyond the first cycle ➔ also,
it is impossible to realize even this filter, but the required filter can be closely
approximated with a smaller time delay.
Practical
Difficulties in
Signal
Reconstruction
(Cont.):

✓ This shows that it is impossible in practice to recover a band-limited


signal g(t) exactly from its samples, even if the sampling rate is higher
than the Nyquist rate.

✓ However, as the sampling rate increases, the recovered signal approaches


the desired signal more closely.
✓ All practical signals are time-limited.
✓ If a signal is time-limited, it cannot be band-limited, and vice versa.
✓ So, all practical signals have infinite bandwidth, and the spectrum has
The Treachery overlapping cycles repeating every fs Hz (the sampling frequency) as shown in
of Aliasing the following figure.
✓ Because of the overlapping tails, the spectrum of sampled signal no longer
has complete information about the original signal, and it is no longer possible
to recover g(t) from the sampled signal.
✓ If the sampled signal is passed through ideal LPF, the output is not G(W) but a
version of G(W) distorted as a result of two separate causes:
▪ The loss of the tail of G(w) beyond f > fs /2 Hz.
▪ The reappearance of this tail inverted or folded onto the spectrum.
✓ The suppression of higher frequencies can be accomplished by an ideal low-
pass filter of bandwidth fs/2 Hz. This filter is called the antialiasing filter and
performed before sampling process.

A Solution: ✓ The antialiasing filter, being an ideal filter, is unrealizable.


Antialiasing Filter
✓ In practice we use a steep cutoff filter, which leaves a sharply attenuated
residual spectrum beyond the folding frequency (fs/2).
Hint
about
BER
Thank you
Dr / Mohamed Youssef
01005150491
[email protected]

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