Sampling

Why SAMPLING?
Sampling plays an essential role in digital communication systems because it
turns continuous analog signals into discrete digital data, allowing them to
be processed, transmitted, stored, and manipulated efficiently in the digital
world. Noise reduction, error detection and correction, compression, signal
processing, and interoperability are all enabled by this conversion, which is
crucial for modern communication systems.turontinuous analog signals into discrete
digital data, allowing them to be processed, transmitted, stored, and manipulated efficiently in the
digital world. Noise reduction, error detection and correction, compression, signal processing, and
interoperability are all enabled by this conversion, which is crucial for modern communication
systems.
Sampling: process of converting CTS to DTS by taking
input signal values of discrete instants of time at
regular or irregular intervals of time
Sampling (Continuous to discrete) is done in a sampler
circuit
• Sample: It is the numeric value of the signal at sampled instant of
time. It is just the signal's measured amplitude at a particular time
and converting it to a digital representation.
• Sampling rate (T) or Sampling frequency (fs): It refers to the number
of samples or data points taken per unit of time. Reciprocal of the
rate is the sampling frequency
• Nyquist rate:
• It is the minimum sampling rate required to accurately capture an
analog signal in digital form without information loss. It is also known
as Nyquist Frequency or Nyquist Limit.
• It is defined as twice the maximum frequency component present in
the analog signal. Fs=2fmax
• Nyquist Interval: The Nyquist interval, also known as the Nyquist
period, is the time interval between consecutive samples in a digital
signal. It is the reciprocal of the Nyquist rate
• T= 1/ Nyquist rate
 • Quantization:
It is the process to represent a continuous-valued signal with a limited
set of discrete values.
OR
In other words, it involves mapping a continuous signal's infinite range
of potential values to a finite collection of discrete values.
 Sampling types
There are three types of sampling techniques:
• Impulse sampling.
• Natural sampling.
• Flat Top sampling.
Impulse sampling : can be performed by multiplying input signal x(t)
with impulse train
Here, the amplitude of impulse changes with respect to amplitude of
input signal x(t)
• This is called ideal sampling or impulse sampling. You cannot use this
practically because pulse width cannot be zero and the generation of
impulse train is not possible practically.
Impulse train sampling

δ T(t) is the periodic impulse train called as sampling function,

T is sampling period and fs is the sampling frequency=2∏/T

Y(t) is an impulse train with the amplitudes

of the impulses equal to the samples of x(t)
At intervals spaced by T

Y(w) is a periodic function of w consisting of

superposition of shifted replicas of X(w)
 Natural Sampling
• Natural sampling is similar to impulse sampling, except the impulse train
is replaced by pulse train of period T.
• multiply input signal x(t) with pulse train
Flat top Sampling
During transmission, noise is introduced at top of the transmission
pulse which can be easily removed if the pulse is in the form of flat
top.
Here, the top of the samples are flat i.e. they have constant
amplitude. Hence, it is called as flat top sampling or practical
sampling.
Flat top sampling makes use of sample and hold circuit.

y(t)=p(t)×yδ(t)......(1)
• Theoretically, the sampled signal can be obtained by
convolution of rectangular pulse p(t) with ideally sampled
signal say yδ(t) as shown in the diagram:
• y(t)=p(t)×yδ(t)......(1)

To get the sampled spectrum, consider Fourier transform on both sides for
equation 1
Y[ω]=F.T[P(t)×yδ(t)]
 Sampling Theorem
• Statement:
A continuous time signal can be represented in its samples and can be
recovered back when sampling frequency fs is greater than or equal to
the twice the highest frequency component of message signal. i. e.
fs≥2fm.
 Nyquist Rate

• It is the minimum sampling rate at which signal can be converted into

samples and can be recovered back without distortion.
• Nyquist rate fN = 2fm hz
• Nyquist interval = 1/fN
= 1/2fm seconds.
Sampling theorem explanation with impulse train
sampling
To reconstruct x(t), input signal spectrum X(ω) must be recovered from
sampled signal spectrum Y(ω), which is possible when there is no
overlapping between the cycles of Y(ω).
Possibility of sampled frequency spectrum with different conditions is given
by the following diagrams

Alliasing
Aliasing and methods to avoid
• It is a phenomenon that occurs when a high-frequency signal is
represented at lower frequency. Means it occurs when the sampling rate
is insufficient and fails to capture the signal properly.
• When the signals are sampled at lower frequency than the nyquist
frequency, high frequency components fold back (gets aliased) in the low
frequency range. This may lead to distorted signal representation.
• In simple words a high frequency component of a signal taking the
identity of low frequency component of a signal when it is
undersampled.
• Methods to Avoid Aliasing
• Sampling at Nyquist Rate
• Using Anti-Aliasing Filter (Low Pass Filter): It helps removing the
component above the nyquist frequency which may lead to aliasing.
UPSAMPLING

L
 Multi-rate Sampling
• Multirate systems have gained popularity since the early 1980s and they
are commonly used for audio and video processing, communications
systems etc.,.
• In most applications multirate systems are used to improve the
performance, or for increased computational efficiency.
• The two basic operations in a multirate system are
• decreasing (decimation) and increasing (interpolation) the
sampling-rate of a signal.
• Multirate systems are sometimes used for sampling-rate conversion,
 Decimation and interpolation
• Decimation can be regarded as the discrete-time counterpart of sampling.
• Whereas in sampling we start with a continuous-time signal x(t) and convert
it into a sequence of samples x[n],
• in decimation we start with a discrete-time signal x[n] and convert it into
another discrete-time signal y[n], which consists of sub-samples of x[n].
• the formal definition of M-fold decimation, or down-sampling, is defined by
• In decimation, the sampling rate is reduced from Fs to Fs/M by discarding M
– 1 samples for every M samples in the original sequence.
• An anti-aliasing digital filter precedes the down-sampler to prevent aliasing
from occurring, due to the lower sampling rate.
• In Figure 9.2 below, it illustrates the concept of 3-fold
• decimation i.e. M = 3. Here, the samples of x[n] corresponding to n = ..., -2,
1, 4,... and n = ..., -1, 2, 5,... are lost in the decimation process. In general,
the samples of x[n] corresponding to n ≠ kM, where k is an integer, are
discarded in M-fold decimation.
• In Figure 9.2 (b), it shows samples of the decimated signal y[n] spaced
three times wider than the
• samples of x[n]. This is not a coincidence. In real time, the decimated signal
appears at a slower rate than that of the
• original signal by a factor of M. If the sampling frequency of x[n] is Fs, then
that of y[n] is Fs/M.
 Interpolation
• Interpolation is the exact opposite of decimation. It is an information
preserving operation, in that all samples of x[n] are
• present in the expanded signal y[n]. The mathematical definition of L-fold
interpolation is defined by Equation 9.2 and
• the block diagram notation is depicted in Figure 9.3. Interpolation works by
inserting (L–1) zero-valued samples for
• each input sample. The sampling rate therefore increases from Fs to LFs.
• Although the expansion process does not cause aliasing in the
interpolated signal, it does however yield undesirable replicas in the
signal’s frequency spectrum.
Sampling rate conversion
• A common use of multirate signal processing is for sampling-rate
conversion.
• Suppose a digital signal x[n] is sampled at an interval T1, and we wish
to obtain a signal y[n] sampled at an interval T2. Then the techniques
of decimation and interpolation enable this operation, providing the
ratio T1/T2 is a rational number i.e. L/M.
• Sampling-rate conversion can be accomplished by L-fold expansion,
followed by low-pass filtering and then M-fold decimation, as shown
in fig . It is important to emphasis that the interpolation should be
performed first and decimation second, to preserve the desired
spectral characteristics of x[n].
• Furthermore by cascading the two in this manner, both of the filters
can be combined into one single low-pass filter.