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Sampling Analog Signal

The document discusses sampling an analog signal to convert it to a digital signal. It describes three key steps: 1) sampling the analog signal at regular intervals, 2) quantizing the sampled signal amplitudes to discrete levels, and 3) encoding the quantized levels into binary digits. It notes that according to the Nyquist theorem, the sampling rate must be at least twice the highest frequency in the analog signal to avoid aliasing when reconstructing the signal.

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Jr Olivarez
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51 views24 pages

Sampling Analog Signal

The document discusses sampling an analog signal to convert it to a digital signal. It describes three key steps: 1) sampling the analog signal at regular intervals, 2) quantizing the sampled signal amplitudes to discrete levels, and 3) encoding the quantized levels into binary digits. It notes that according to the Nyquist theorem, the sampling rate must be at least twice the highest frequency in the analog signal to avoid aliasing when reconstructing the signal.

Uploaded by

Jr Olivarez
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Sampling Analog signal

ANALOG-TO-DIGITAL CONVERSION

A digital signal is superior to an analog signal because


it is more robust to noise and can easily be recovered,
corrected and amplified. For this reason, the tendency
today is to change an analog signal to digital data.

4.2
PCM
• PCM consists of three steps to digitize an
analog signal:
1. Sampling
2. Quantization
3. Binary encoding
 Before we sample, we have to filter the
signal to limit the maximum frequency of the
signal as it affects the sampling rate.
 Filtering should ensure that we do not
distort the signal, ie remove high frequency
components that affect the signal shape.

4.3
Figure 4.21 Components of PCM encoder

4.4
Sampling
• Analog signal is sampled every TS secs.
• Ts is referred to as the sampling interval.
• fs = 1/Ts is called the sampling rate or
sampling frequency.
• There are 3 sampling methods:
– Ideal - an impulse at each sampling instant
– Natural - a pulse of short width with varying
amplitude
– Flattop - sample and hold, like natural but with
single amplitude value
• The process is referred to as pulse amplitude
modulation PAM and the outcome is a signal
with analog (non integer) values
4.5
Figure 4.22 Three different sampling methods for PCM

4.6
Note

According to the Nyquist theorem, the


sampling rate must be
at least 2 times the highest frequency
contained in the signal.

4.7
Figure 4.23 Nyquist sampling rate for low-pass and bandpass signals

4.8
Example 4.6
For an intuitive example of the Nyquist theorem, let us
sample a simple sine wave at three sampling rates: fs =
4f (2 times the Nyquist rate), fs = 2f (Nyquist rate), and
fs = f (one-half the Nyquist rate). Figure 4.24 shows the
sampling and the subsequent recovery of the signal.

It can be seen that sampling at the Nyquist rate can


create a good approximation of the original sine wave
(part a). Oversampling in part b can also create the
same approximation, but it is redundant and
unnecessary. Sampling below the Nyquist rate (part c)
does not produce a signal that looks like the original
sine wave.

4.9
Figure 4.24 Recovery of a sampled sine wave for different sampling rates

4.10
Aliasing

• An effect that causes different signals to


become indistinguishable (oraliases of one
another) when sampled. It also refers to
the distortion or artifact that results when
the signal reconstructed from samples is
different from the original continuous
signal.
Example
Example
Example

4.15
Example

A complex low-pass signal has a bandwidth of 200 kHz.


What is the minimum sampling rate for this signal?

Solution
The bandwidth of a low-pass signal is between 0 and f,
where f is the maximum frequency in the signal.
Therefore, we can sample this signal at 2 times the
highest frequency (200 kHz). The sampling rate is
therefore 400,000 samples per second.

4.16
Example

A complex bandpass signal has a bandwidth of 200 kHz.


What is the minimum sampling rate for this signal?

Solution
We cannot find the minimum sampling rate in this case
because we do not know where the bandwidth starts or
ends. We do not know the maximum frequency in the
signal.

4.17
Quantization
• Sampling results in a series of pulses of
varying amplitude values ranging between
two limits: a min and a max.
• The amplitude values are infinite between the
two limits.
• We need to map the infinite amplitude values
onto a finite set of known values.
• This is achieved by dividing the distance
between min and max into L zones, each of
height 
 = (max - min)/L

4.18
Quantization Levels

• The midpoint of each zone is assigned


a value from 0 to L-1 (resulting in L
values)
• Each sample falling in a zone is then
approximated to the value of the
midpoint.

4.19
Quantization Zones
• Assume we have a voltage signal with
amplitutes Vmin=-20V and Vmax=+20V.
• We want to use L=8 quantization levels.
• Zone width = (20 - -20)/8 = 5
• The 8 zones are: -20 to -15, -15 to -10, -
10 to -5, -5 to 0, 0 to +5, +5 to +10, +10
to +15, +15 to +20
• The midpoints are: -17.5, -12.5, -7.5, -
2.5, 2.5, 7.5, 12.5, 17.5
4.20
Assigning Codes to Zones
• Each zone is then assigned a binary code.
• The number of bits required to encode the
zones, or the number of bits per sample as it is
commonly referred to, is obtained as follows:
nb = log2 L
• Given our example, nb = 3
• The 8 zone (or level) codes are therefore: 000,
001, 010, 011, 100, 101, 110, and 111
• Assigning codes to zones:
– 000 will refer to zone -20 to -15
– 001 to zone -15 to -10, etc.

4.21
Figure 4.26 Quantization and encoding of a sampled signal

4.22
Bit rate and bandwidth
requirements of PCM
• The bit rate of a PCM signal can be calculated form
the number of bits per sample x the sampling rate
Bit rate = nb x fs
• The bandwidth required to transmit this signal
depends on the type of line encoding used. Refer to
previous section for discussion and formulas.
• A digitized signal will always need more bandwidth
than the original analog signal. Price we pay for
robustness and other features of digital transmission.

4.23
Example

We want to digitize the human voice. What is the bit rate,


assuming 8 bits per sample?

Solution
The human voice normally contains frequencies from 0
to 4000 Hz. So the sampling rate and bit rate are
calculated as follows:

4.24

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