Unit 4

Nyquist Sampling Theorem

and Pulse Amplitude

Taught by:
Dr. Prateek Verma
Assistant Professor
DYPIU Akurdi
1
 Introduction to Discrete Time Signals
• There are two types of signals – Continuous time signal and Discrete time signal.

• During the recent advancement in the field, discrete time systems are preferred due to
inexpensive, light weight etc.

• Therefore, discrete time signals are more preferred and processed for processing of
continuous time signals.

2
 Contd.
• Although in real world, most of the signals generated are in continuous time signals but
they are converted to discrete time signals for processing.

• This conversion from continuous time to discrete time signals is done by sampling
theorem.

• With the help of sampling theorem, a continuous time signal may be completely
recovered from the knowledge of samples taken uniformly.

• Therefore, sampling theorem can be seen as a bridge between continuous time and
discrete time signals.

3
 The Sampling Theorem
• A continuous time signal is first converted into discrete time signals by sampling
process.

• The sufficient number of samples of the signal must be taken so that the original signal is
represented by the samples completely.

• Also, it should be possible to recover or reconstruct the original signal completely from
its samples.

• The number of samples to be taken depends upon the maximum signal frequency present
in the signal.

• Sampling theorem provides this information about the number of samples. 4

 The Sampling Theorem
• A continuous time signal may be completely represented in its samples and recovered
back if the sampling frequency is 𝒇𝒔 ≥ 𝟐𝒇𝒎 .

• Here, 𝑓𝑠 is the sampling frequency and 𝑓𝑚 is the maximum frequency present in the
signal.

• The statement of sampling theorem can also be given as:

• A band limited signal of finite energy, which has no frequency components higher than 𝑓𝑚 Hz, is
completely described by its sample values at uniform intervals less than or equal to 1/2𝑓𝑚 sec apart.

• A band limited signal of finite energy, which has no frequency components higher than 𝑓𝑚 Hz, may be
completely recovered from the knowledge of its samples taken at the rate of 2𝑓𝑚 samples per second.

5
 Proof of the Sampling Theorem
• Let us consider a continuous time band limited signal to 𝑓𝑚 Hz, which can be
reconstructed exactly without any error from its samples taken uniformly at a rate 𝑓𝑠 >
2𝑓𝑚 Hz.

• A band-limited signal means its Fourier Transform or its spectrum will be non-zero only
within a small range of frequencies.
𝑀(𝜔)

FT

−𝜔𝑚 𝜔𝑚 𝜔
0

6
 Contd.
• We will feed the message signal to a sampler (which acts as a multiplier). The first signal
is 𝑥(𝑡) whose max. freq. component is ω𝑚 and second signal is 𝛿𝑇𝑠 (𝑡).

• Here, 𝛿𝑇𝑠 𝑡 is a periodic impulse train having fundamental time period 𝑇𝑠 (also called as
sampling period or sampling interval.
𝑔 𝑡 = x t . c(t)

Sampler

𝑐 𝑡 = 𝛿(𝑡 − 𝑛𝑇𝑠 )
2𝜋 𝑛=−∞
𝜔𝑠 = = 𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑓𝑟𝑒𝑞.
𝑇𝑠 7
 Contd.
• Here, the weight of each impulse is equal to the instantaneous value of the message
signal.

• Next step is to obtain the spectrum of the sampled signal. FT

g(t) 𝐺(𝜔)

Since g(t) = x(t).c(t) 𝑋(𝜔)

FT

−𝜔𝑚 0 𝜔𝑚 𝜔
1
G(𝜔) = [𝑋 𝜔 ∗𝐶 𝜔 ]
2𝜋 ∞

𝐶 𝜔 = 𝜔𝑠 𝛿(𝜔 − 𝑛𝜔𝑠 )
Convolution of freq. 𝑛=−∞
Sampled signal spectrum of two signals 8
 Contd.
• For simplification, we will use property of convolution and properties of impulse function.

∞ ∞
1 𝜔𝑠
𝐺(𝜔) = [𝑋 𝜔 ∗ 𝜔𝑠 𝛿(𝜔 − 𝑛𝜔𝑠 )] 𝐺(𝜔) = [𝑋 𝜔 ∗ 𝛿(𝜔 − 𝑛𝜔𝑠 )]
2𝜋 2𝜋
𝑛=−∞ 𝑛=−∞

∞
• Now we will use one property, 1
𝐺(𝜔) = [ 𝑋 𝜔 ∗ 𝛿(𝜔 − 𝑛𝜔𝑠 )]
𝑇𝑠
𝑛=−∞
𝑥 𝑡 ∗ 𝛿 𝑡 − 𝑡1 = 𝑥(𝑡 − 𝑡1 )
∞
1
• Therefore, 𝐺(𝜔) will become 𝐺(𝜔) = 𝑋 𝜔 − 𝑛𝜔𝑠
𝑇𝑠
𝑛=−∞

𝟏
𝑮(𝝎) = [… + 𝑿 𝝎 + 𝝎𝒔 + 𝑿 𝝎 + 𝑿 𝝎 − 𝝎𝒔 + ⋯ ]
𝑻𝒔
9
 Contd.
• Now we will plot the wave for this equation 𝟏
𝑮(𝝎) = [… + 𝑿 𝝎 + 𝝎𝒔 + 𝑿 𝝎 + 𝑿 𝝎 − 𝝎𝒔 + ⋯ ]
𝑻𝒔
𝑮(𝝎)
𝜔𝑠 − 𝜔𝑚 > 𝜔𝑚

−𝜔m 0 +𝜔m 𝝎 𝜔𝑠 > 2𝜔𝑚

𝜔𝑠 − 𝜔m 𝜔𝑠 + 𝜔m

• If 𝑓𝑠 > 2𝑓𝑚 , the spectrum 𝐺(𝜔) will repeat periodically without overlapping.

• The spectrum of sampled signal extends up to infinity and ideal bandwidth of sampled signal is
infinite. But we have to extract original spectrum X(𝜔) from 𝐺(𝜔).
10
 Contd.
• The original or desired spectrum X(𝜔) is centered at 𝜔 = 0 and is having bandwidth or
maximum frequency equal to ωm .

• The desired spectrum can be obtained by passing the sampled signal through a Low Pass Filter
(LPF) with cut off frequency ωm .

• The extracted signal X(𝜔) can then be converted into time domain signal x(𝑡).

𝑿(𝝎)
𝑮(𝝎)
LPF

−𝜔m 0 +𝜔m 𝝎
−𝜔m 0 +𝜔m 𝝎
𝜔𝑠 − 𝜔m 𝜔𝑠 + 𝜔m 𝑩𝑾 = 𝟐𝝎𝒎 11
 Contd.
• For the case 𝑓𝑠 = 2𝑓𝑚 , the successive cycles of 𝐺 𝜔 are not overlapping with each other but
they are touching each other. In this case also, we can recover the original signal.
𝑮(𝝎)
𝑿(𝝎)
LPF

−𝜔m 0 +𝜔m 𝝎 −𝜔m 0 +𝜔m 𝝎

• For the case 𝑓𝑠 < 2𝑓𝑚 , the successive cycles of 𝐺 𝜔 will overlap with each other and hence
original spectrum of 𝑋 𝜔 cannot be extracted from 𝐺 𝜔 .
𝑮(𝝎)

𝝎 12
 Nyquist Rate and Nyquist Interval
• When the sampling rate becomes exactly equal to 2𝑓𝑚 samples per second, then it is called
Nyquist Rate.

• Nyquist Rate is also called as the minimum sampling rate.

• It is given by 𝒇𝒔 = 𝟐𝒇𝒎 .

• Similarly, maximum time interval between equally spaced samples is called as Nyquist Interval.
It is given by
𝑮(𝝎)
1
𝑁𝑦𝑞𝑢𝑖𝑠𝑡 𝐼𝑛𝑡𝑒𝑟𝑣𝑎𝑙 = 𝑇𝑠 = seconds
2𝑓𝑚

13
−𝜔m 0 +𝜔m 𝝎
 Reconstruction Filter (LPF)
• The LPF is used to reconstruct original signal from its samples. LPF will pass only low
frequencies upto a specified cutoff frequency.

• Here, the frequency response of an ideal LPF is shown, which shows there is a sharp change in
response at cut off frequency (amplitude suddenly becomes zero which is not possible
practically).

• This means that an ideal low pass filter is not

physically realizable.

• Therefore, we use practical low pass filter for

extracting original signal 𝑋(𝜔).

14
 Practical LPF
• In case of practical LPF, the amplitude response decreases slowly to become zero.

• Practically, realizable filters require a non-zero bandwidth for the transition between the
passband and the required out of band attenuation.

• This means that there is a transition band in case of practical filter.

• To minimize the sample rate, we want the small transition

bandwidth.

• Filter complexity and cost rise sharply with narrower

transition bandwidth, so a trade off is required between cost
of a small transition bandwidth and cost of a high sampling
rate. 15
Problem during reconstruction
𝑮(𝝎)

Not Possible

−𝜔m 0 +𝜔m 𝝎

𝑮(𝝎)

Possible
−𝜔m 0 +𝜔m 𝝎

16
 Signal Reconstruction: The Interpolation Process
• The process of reconstructing a continuous-time signal 𝑥(𝑡) from its samples is called as
Interpolation.

• A signal 𝑥(𝑡) band-limited to 𝑓𝑚 Hz can be reconstructed (interpolated) completely from its

samples.

• This is achieved by passing the sampled signal through an ideal low-pass filter of cut-off
frequency 𝑓𝑚 Hz.

17
 Sampling Techniques
• There are three types of sampling techniques as under:

Instantaneous Natural Flat Top

sampling sampling sampling

• Instantaneous sampling is called ideal sampling while natural and Flat Top sampling are called
practical sampling methods.

18
 Instantaneous/Ideal/Impulse sampling
• In this, the sampling function is a train of impulses.
∞

𝐶 𝜔 = 𝜔𝑠 𝛿(𝜔 − 𝑛𝜔𝑠 )
𝑛=−∞

𝛿𝑇𝑠 𝑡 = 𝛿(𝑡 − 𝑛𝑇𝑠 )

𝑛=−∞

• It may be noted that ideal sampling is possible only in theory since it is impossible to have a
pulse whose width approached zero.

• Practically flat top sampling and nature sampling are used.

19
 Natural sampling
• In this sampling, the pulse has a finite width equal to 𝜏.

• In this method, a sampled signal 𝑔 𝑡 = 𝑥 𝑡 . 𝑐 𝑡 (1)

• Here, 𝑐 𝑡 is the periodic pulse of width 𝜏 and freq. 𝑓𝑠 .

• When c 𝑡 goes high, the switch ‘S’ is closed.

• Therefore, 𝑔 𝑡 = 𝑥(𝑡) when 𝑐 𝑡 = 𝐴 (amplitude of 𝑐 𝑡 )

𝑔 𝑡 =0 when 𝑐 𝑡 = 0 20
Contd.
• The exponential FS for a periodic waveform is
expressed as
∞

𝑐 𝑡 = 𝐶𝑛 𝑒 𝑗2𝜋𝑛𝑡/𝑇0 (2)
𝑛=−∞

• Here, 𝑇0 = 𝑇𝑠 = 1/𝑓𝑠 (period of 𝑐 𝑡 ) or

𝑓0 = 𝑓𝑠 = 1/𝑇0

• Therefore, Equation 2 can be written as:

∞

𝑐 𝑡 = 𝐶𝑛 𝑒 𝑗2𝜋𝑛𝑓𝑠 𝑡 (3)
𝑛=−∞
21
Contd.
• Now, 𝑐 𝑡 is a rectangular pulse train, so 𝐶𝑛 will be
𝜏𝐴
𝐶𝑛 = 𝑠𝑖𝑛𝑐(𝑓𝑛 𝜏) (4)
𝑇𝑠
∞
𝜏𝐴
• Using Equation 3 and 4, the FS in Eq. 2 can be represented as 𝑐 𝑡 = 𝑠𝑖𝑛𝑐 𝑓𝑛 𝜏 𝑒 𝑗2𝜋𝑓𝑠 𝑛𝑡
𝑇𝑠 (5)
𝑛=−∞

• Now putting this value of 𝑐 𝑡 from Equation 5 to Equation 1

∞
𝝉𝑨
𝑔 𝑡 = 𝑥 𝑡 .𝑐 𝑡 𝒈 𝒕 = 𝒔𝒊𝒏𝒄 𝒇𝒏 𝝉 𝒆𝒋𝟐𝝅𝒇𝒔𝒏𝒕 . 𝒙(𝒕) (6)
𝑻𝒔
𝒏=−∞

• This is the required time-domain representation for naturally sampled signal.

22
Contd.
• Now, we have to find the frequency domain representation of the signal 𝑔(𝑡) by taking the
Fourier Transform of the signal.
∞
𝜏𝐴
𝐺 𝑓 = 𝐹𝑇 𝑔 𝑡 = 𝑠𝑖𝑛𝑐 𝑓𝑛 𝜏 𝐹𝑇[𝑒 𝑗2𝜋𝑓𝑠 𝑛𝑡 . 𝑥 𝑡 ] (7)
𝑇𝑠
𝑛=−∞

• Now using the Frequency shifting property of FT 𝑒 𝑗2𝜋𝑓𝑠 𝑛𝑡 . 𝑥 𝑡

FT
𝑋(𝑓 − 𝑓𝑠 𝑛)

∞
• Therefore, 𝐺(𝑓) will become 𝝉𝑨
𝑮 𝒇 = 𝒔𝒊𝒏𝒄 𝒇𝒏 𝝉 𝑿 𝒇 − 𝒇𝒔 𝒏 (8)
𝑻𝒔
𝒏=−∞

• Equation 8 suggests that the spectra of 𝑥(𝑡) i.e. 𝑋(𝑓) is periodic in nature with freq. 𝑓𝑠 and is
weighed by the sinc function.

23
Contd.

∞
𝝉𝑨
𝑮 𝒇 = 𝒔𝒊𝒏𝒄 𝒇𝒏 𝝉 𝑿 𝒇 − 𝒇𝒔 𝒏
𝑻𝒔
𝒏=−∞

• In ideal sampling, the spectrum remain constant throughout the frequency range while in
natural sampling the spectrum is weighted by a sinc function.

24
Flat Top sampling or Rectangular Pulse sampling
• Flat Top sampling is similar to Natural sampling but natural sampling is little complex whereas it is
easy to get flat top sampling.

• In Flat Top sampling, the top of the samples remains constant and is equal to the instantaneous
value of the baseband signal at the start of the sampling.

• The starting edge of the pulse represents the

point where the message signal is sampled.

• The duration or width of each sample is 𝜏 and

1
sampling rate is equal to 𝑓𝑠 = .
𝑇𝑠

25
Tutorial
• An analog signal is expressed by the equation 𝑥 𝑡 = 3 cos 50𝜋𝑡 + 10 sin 300𝜋𝑡 −
cos(100𝜋𝑡). Calculate the Nyquist rate for this signal.

Ans – Nyquist rate – 300 Hz

1
• Find the Nyquist rate and the Nyquist Interval for the signal 𝑥 𝑡 = cos 4000𝜋𝑡 cos 1000𝜋𝑡 .
2𝜋

Ans – Hint Use 2cosAcosB = cos (A+B)+cos(A-B)

Nyquist rate – 5 KHz Nyquist Interval – 0.2 msec

26
Tutorial
• Determine the Nyquist rate of a continuous time signal 𝑥 𝑡 = 6𝑐𝑜𝑠50𝜋𝑡 + 20𝑠𝑖𝑛300𝜋𝑡 −
10𝑐𝑜𝑠100𝜋𝑡.

Ans – Nyquist rate – 300 Hz

sin 1000𝜋𝑡
• Determine the Nyquist rate of a continuous time signal 𝑥 𝑡 = . 𝜋
𝜋𝑡

𝑠𝑖𝑛𝑥 𝑊 FT 𝑟𝑒𝑐𝑡 𝜔/2𝑊 𝜔

𝑠𝑖𝑛𝑐 𝑥 = 𝑠𝑖𝑛𝑐 𝑊𝑡 −1000𝜋 1000𝜋
𝑥 𝜋
𝜔𝑚𝑎𝑥 = 1000𝜋
sin 1000𝜋𝑡 1000 sin 1000𝜋𝑡 = 1000𝑠𝑖𝑛𝑐(1000𝜋𝑡)
𝑥 𝑡 = = 𝑓𝑚𝑎𝑥 = 500 Hz
𝜋𝑡 1000𝜋𝑡
𝜋1000 𝜔 𝑓𝑠 = 2𝑓𝑚 = 1000 Hz
𝑠𝑖𝑛𝑐(1000𝜋𝑡) FT 𝜋𝑟𝑒𝑐𝑡 27
𝜋 2(1000𝜋)
Aperture Effect
• Aperture effect is observed in Flat Top Sampling.

• As the duration ‘τ’ of the pulse increases, the aperture effect is more prominent. Hence,
during reconstruction an equalizer is needed to compensate for this effect.

• The distortion caused by the use of PAM to

transmit an analog information bearing signal
is referred to as the aperture effect.

• The width of pulse can be changed in order to

reduce aperture effect.

28
Analog Pulse Modulation Methods
• In Pulse modulation methods, the carrier is no longer a continuous signal but consists of a
pulse train.

• Some parameter of this pulse train is varied according to the instantaneous value of the
modulating signal.

• There are two types of pulse modulation systems as under:

Pulse Amplitude Pulse Time

Modulation Modulation

29
Contd.
• In PAM, amplitude of the pulses
of the carrier is varied in
accordance with the modulating
signal.

• In PTM, the timing of the pulses

of the carrier is varied according
to the instantaneous value of the
modulating signal.

• There are two types of PTM:

Pulse Pulse
Width Position
30
Contd.
• In PWM, width of the pulses of
the carrier is varied in accordance
with the modulating signal.

• PWM is also called as Pulse

Duration modulation.

• In PPM, the position of the pulses

of the carrier is varied according
to the instantaneous value of the
modulating signal.

31
Pulse Amplitude Modulation (PAM)
• Generally, the Flat Top PAM method is preferred and widely used.

• The reason for that is during the transmission, the noise interferes with the top of the
transmitted pulses and this noise can be easily removed if the PAM pulse has flat top.

• However, in case of natural sampled PAM

signal, the pulse has varying top in
accordance with the message signal.

• When noise effects the signal, then it

becomes quite difficult to remove the
noise and detect the original signal.

• Hence errors are introduced in the signal.

32
Contd.
• A sample and hold circuit is used to produce a Flat Top sampled PAM signal.

• G1 (Sampling switch) and G2 (Discharge switch) are the field effect transistors (FET)
with a capacitor C.

• G1 is closed for a short duration by a short

pulse applied to G1 transistor.

• During this time, the capacitor is fully

charged upto a voltage equal to the
instantaneous sample value of message
signal.

• Now sampling switch is opened and

Capacitor holds the charge. 33
Contd.
• The discharge switch is then closed by a pulse applied to G2 and in that duration capacitor
C discharged to 0.

• The bandwidth required to transmit a PAM signal is equal to maximum freq.

1
• 𝐵𝑊 ≥ 𝑓𝑚𝑎𝑥 and 𝑓𝑚𝑎𝑥 =
2𝜏

𝟏
• Therefore, 𝑩𝑾 ≥
𝟐𝝉

• For a PAM voice signal having maximum frequency as 3 KHz, calculate the transmission
bandwidth. It is given that sampling frequency is 8 KHz and pulse duration 𝜏 = 0.1𝑇𝑠 .

Ans - 𝐵𝑊 ≥ 40𝐾𝐻𝑧 34
 Drawbacks of PAM signal
• The bandwidth required to transmit a PAM signal is very large in comparison to the
maximum freq. present in the message signal.

• Since amplitude is varied in PAM signal, therefore interference of noise is maximum.

35
 PWM or PDM
• The width of the pulse changes with the message signal.

• Therefore, information is contained in the width variation.

• Since the noise changes the amplitude of the signal, therefore PWM system is more immune
to noise than PAM.

• The disadvantages of PWM signal is:

• Bandwidth required for PWM transmission is large as compared to the bandwidth of PAM signal.
• Due to variable pulse width, the pulses have variable power contents.

36
Contd.

37
PPM
• Since the noise changes the amplitude of the signal, therefore PPM system is more
immune to noise than PAM.

• Due to constant amplitude of pulses, the transmitted power always remain constant. It
does not change as the case in PWM.

• The disadvantages of PWM signal is:

• Large Bandwidth is required to ensure proper transmission for PWM signal.
• Since the position of the pulse is required to recover the original signal, a synchronizing pulse is
required for a timing circuit in the receiver. The demodulation is not possible without them.

38
Comparison of PAM, PWM and PPM

39
40