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Digital Communication: Sampling

Digital communication systems use sampling to convert analog signals to digital form for transmission. Sampling involves taking periodic measurements of the amplitude of an analog signal. This results in a discrete-time representation of the original signal. The sampling rate must be at least twice the bandwidth of the original signal as per Nyquist's theorem to avoid aliasing, where frequency components above the Nyquist rate are incorrectly interpreted. Undersampling below this rate causes aliasing distortion during reconstruction.

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15 views14 pages

Digital Communication: Sampling

Digital communication systems use sampling to convert analog signals to digital form for transmission. Sampling involves taking periodic measurements of the amplitude of an analog signal. This results in a discrete-time representation of the original signal. The sampling rate must be at least twice the bandwidth of the original signal as per Nyquist's theorem to avoid aliasing, where frequency components above the Nyquist rate are incorrectly interpreted. Undersampling below this rate causes aliasing distortion during reconstruction.

Uploaded by

rkishore
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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Digital Communication

SAMPLING

Sampling
The input signal is sampled prior to digitisation and an approximation to the input is reconstructed by the digital-toanalogue converter:
input

Sampling

Digitisation

code, modulate
Transmission Wire/optical fibre Aerial/free-space

Filtering

Digital-to-analogue

conversion

Demodulate, Decode

output

Sampling an analogue signal


Prior to digitisation, signals must be sampled
ADC converts the height of each pulse into binary representation Sampling involves the multiplication of the signal by a train of sampling pulses
With a frequency fs=2B=1/T

Sampling as multiplication by a sampling waveform:

Sampling pulse is short enough so that can normally considered have zero duration DAC, however produces pulses length T

Multiplication = Amplitude modulation


Amplitude modulation produces sidebands

Sidebands produced by multiplication with a carrier


That is, amplitude modulation

Sidebands at each harmonic of the sampling pulse Digital-to-analogue conversion involves recovery of the baseband
How? What is the minimum value of fs for which there is no overlap of the Harmonics with the baseband?

If the sidebands do not overlap the signal can be recovered

Practical sampling
the "Sample-and-hold" system:

This is Nyquists theorem


For a signal of bandwidth B Hz, the minimum sampling rate is 2B samples/s

Effect of sampling rate


sampling at more than the Nyquist Rate

Sampling at the Nyquist Rate


cannot build an ideal filter -

Undersampling
produces aliasing distortion!

Aliasing-time domain

Oversampled signal

Reconstructed signal

Undersampled signal

Reconstructed signal Sampling:aliasing & Nyquist:time domain

The Anti-alias (Pre-sampling) filter


ensures that sampling obeys the Nyquist theorem

Examples
For the compact disc (Audio CD) the maximum signal frequency is 20 kHz and the sampling rate is 44.1 kHz.
The Nyquist Sampling Rate is 40 kHz Hence the guard band is 4.1 kHz wide.

In the telephone system (see Section 5.8), the speech signal has a bandwidth up to 3.4 kHz and a sampling rate of 8 kHz,
The Nyquist Sampling Rate is 6.8 kHz Hence the guard band is 1.2 kHz wide.

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