The lab focuses on sampling audio signals while addressing aliasing, emphasizing the importance of the Nyquist sampling theorem, which states that the sampling frequency must be at least twice the highest frequency in the signal to avoid distortion. It includes tasks to prove the theorem by sampling specific waveforms at various rates and observing the effects of under-sampling and over-sampling. The document outlines the theoretical background and practical applications of sampling in signal processing.
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Digital Signal Processing LAB
The lab focuses on sampling audio signals while addressing aliasing, emphasizing the importance of the Nyquist sampling theorem, which states that the sampling frequency must be at least twice the highest frequency in the signal to avoid distortion. It includes tasks to prove the theorem by sampling specific waveforms at various rates and observing the effects of under-sampling and over-sampling. The document outlines the theoretical background and practical applications of sampling in signal processing.
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Lab # 09: Sampling of Audio Signals and Aliasing
Objectives:
The objective of this lab is to perform sampling on audio signals while taking care of aliasing,
Descriptio
In signal processing, sampling is the reduction of a continuous signal to a discrete signal. A
common example is the conversion of a sound wave (a continuous signal) to a sequence of samples
(2 discrete-time signal),
A sample is a value or set of values at a point in time and/or space. While a sampler is a subsystem
or operation that extracts samples from a continuous signal.
A theoretical ideal sampler produces samples equivalent to the instantaneous value of the
continuous signal at the desired points.
‘The Nyquist sampling theorem provides a prescription for the nominal sampling interval required
to avoid aliasing. It may be stated simply as follows:
The sampling frequency should be at least twice the highest frequency contained in the signal.
F,>=2F.
where fs is the sampling frequency (how often samples are taken per unit of time or space), and fe
is the highest frequency contained in the signal. That this is so is really quite intuitive, Consider
for example a signal composed of a single sinewave at a frequency of 1 Hz:
2
1
time (see)
If we sample this waveform at 2 Hz (as dictated by the Nyquist theorem), that is sufficient to
capture each peak and trough of the signal:
40 time (580)
If we sample at a frequency higher than this, for example 3 Hz, then there are more than enough
samples to capture the variations in the signal:
2
AAAS
amps
0 os. + 45. 2 38 3 a5. 4
time se)
If, however we sample at a frequency lower than 2 Hz, for example at 1.5 Hz, then there are now
not enough samples to capture all the peaks and troughs in the signal:
i We V7 NS Xd
mete)
41 Note here that we are not only losing information, but we are getting the wrong information about
the signal. The person receiving these samples, without any previous knowledge of the original
signal, may well be misled into thinking that the signal has quite a different form:
8 '
aq 1 r r
§ 4 PK
4 Y
-13|
ay Ds 1 15 2 25 3 35 4
time (se)
From this example, we can see the reason for the term aliasing. That is, the signal now takes on a
different \persona," or a false presentation, due to being sampled at an insufficiently high
frequency. Now we are ready to think about the sampling of a complex signal composed of many
frequency cone oe theorem, we know that any continuous signal may be
decomposed in terms of a sunVof sines and cosines at different frequencies,
For example, the following waveform was composed by adding together sine waves at frequencies
of 1 Hz, 2 Hz, and 3 Hz:
00s CS
sme ao) |
According to the Nyquist sampling theorem, the signal must be sampled at twice the highest
frequency contained in the signal. In this case, we have ind so the Nyquist theorem tells
us that the sampling frequency, fs, must be at leaSf 6 Hz, sure enough, this appears to be
sufficient:
42 LAB TASK:
1. Prove Nyquist’s Sampling Theorem, by sampling the following waves
a) Y= Cos(2*pi* it)
Where #10Hz,
b) Y =Sin(2*pi*f*t) + Cos(2*pi*f2*t)
Where fl = 50Hz.and £2 = 200Hz
at three possible sampling rates i.e.
(1) Fy =2Fe (2) Fe<2Fe (3) Fy>2Fe
2. Sample an audio with Fs=2Fe, Fs < 2Fe and Fs >2Fe. Observe the effect of all cases and plot
signal
43
