Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

UNIT
Sampling, PCM, DPCM, DM
& TDM
SYLLABUS
Pulse modulation, Sampling Process, Pulse Amplitude and Pulse code modulation (PCM),
Differential Pulse Code Modulation (DPCM), Delta Modulation (DM), Noise considerations in
PCM, Time Division Multiplexing (TDM), Digital Multiplexers.

Sampling:
Sampling is a process where an analog signal is converted into a corresponding sequence of
samples that are usually spaced uniformly in time. i.e., Process of Converting continuous time
signal into discrete time signal.
There are two types of Sampling:
1. Ideal Sampling or Impulse Sampling or Instantaneous Sampling
2. Practical Sampling.
i. Natural Sampling or chopper Sampling
ii. Flat Top Sampling or sample & Hold Sampling.

Ideal Sampling, or Impulse Sampling or Instantaneous Sampling:

Statement:
"A band-limited Signal having frequency 'W' Hz can be completely described by its Samples if the
Sampling Frequency fs is greater than or equal to twice the highest frequency of message signal."

A band-limited Signal having the highest frequency ' W ' Hz can be completely Recovered from its
Samples if the Samples are taken at the rate 'fs' greater than or equal to twice the highest
frequency of message signal, i.e. fs ≥ 2W

Fig. 1: Sampling
The Sampled Signal g⸹(t) is obtained by multiplying the analog signal g(t) by a sequence of
Impulse Sδ(t) which is periodic with period Ts seconds.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Time-domain analysis:

Fig. 2: Sampling Waveform

Sampled output g⸹(t) is given by.

𝑔∂ (t) = g(t)⋅ 𝑆∂ (𝑡) → (1)
periodic Impulse train S⸹(t) can be represented as:
∞

𝑆∂ (𝑡) = ∑ ∂(𝑡 − 𝑛𝑇𝑠 ) → (2)

𝑛=−∞
Substituting eq (2) in eq (1), we get
∞

𝑔𝛿 (𝑡) = 𝑔(𝑡). ∑ ∂(𝑡 − 𝑛𝑇𝑠 )

𝑛=−∞
using the shifting property of the impulse function
𝑔(𝑡) ⋅ ∂(𝑡 − 𝑛𝑇𝑠 ) = 𝑔(𝑛𝑇𝑠 ) ∂(𝑡 − 𝑛𝑇𝑠 )
∞

∴ 𝑔∂ (t) = ∑ 𝑔(𝑛𝑇𝑠 ) ∂(t − 𝑛𝑇𝑠 )

𝑛=−∞

Frequency domain analysis (Spectrum analysis):

Taking Fourier transform of Eq. (1), we get
𝐺𝜕 (𝑓) = 𝐺(𝑓) ∗ 𝑆𝜕 (𝑓) → (4)
∞

Where 𝑆𝜕 (𝑓) = 𝑓𝑠 ∑ ∂(𝑓 − 𝑛𝑓𝑠 ) → (5)

𝑛=−∞
Substituting eq (5) in eq (4), we get
∞

𝐺𝜕 (𝑓) = 𝐺(𝑓) ∗ 𝑓𝑠 ∑ ∂(𝑓 − 𝑛𝑓𝑠 )

𝑛=−∞
From the convolution property of the impulse function
𝐺(𝑓) ∗ ∂(𝑓 − 𝑛𝑓𝑠 ) = 𝐺(𝑓 − 𝑛𝑓𝑠 )
∞

𝐺𝜕 (𝑓) = 𝑓𝑠 ∑ 𝐺(𝑓 − 𝑛𝑓𝑠 ) → (6)

𝑛=−∞

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Equation (6) can be written as

∞

𝐺𝜕 (𝑓) = 𝑓𝑠 . 𝐺(𝑓) + 𝑓𝑠 ∑ 𝐺(𝑓 − 𝑛𝑓𝑠 ) → (7)

𝑛=−∞
When the spectrum of 𝐺∂ (𝑓) i.e. eq (7) is passed through an LPF, then the 2nd terms in R.H.S of
Eq. (7) is eliminated, resulting in
𝐺𝜕 (𝑓) = 𝑓𝑆 𝐺(𝑓)
1
∴ 𝐺(𝐹) = 𝐺𝜕 (𝑓) → (8)
𝑓𝑆

𝐋𝐞𝐭 𝒇𝑺 = 𝟐𝑾
Another useful expression for the 𝐹𝑇 of 𝐺∂ (𝑓) may be obtained by taking the FT on both side of
Eq. (3)
∞

FT {𝑔𝜕 (𝑡) = ∑ 𝑔(𝑛𝑇𝑠 ) ∂(𝑡 − 𝑛𝑇𝑠 )}

𝑛=−∞

Wkt
𝑭𝑻
𝛛(𝒕 − 𝒏𝑻𝒔 ) ⟶ 𝒆−𝒋𝟐𝝅𝒏𝒇𝑻𝒔

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM
∞

𝐺𝜕 (𝑓) = ∑ 𝑔(𝑛𝑇𝑠 )𝑒 −𝑗2𝜋𝑛𝑓𝑇𝑠 → (9)

𝑛=−∞

Substituting eq (9) in eq (8) & putting 𝒇𝑺 = 𝟐𝑾

∞
1 𝑛 −𝑗2𝜋𝑛𝑓
𝐺(𝑓) = ∑ 𝑔 ( ) 𝑒 2𝑊
2𝑊 2𝑤
𝑛=−∞

∞
𝟏 𝒏 −𝒋𝝅𝒏𝒇
𝑮(𝒇) = ∑ 𝒈 ( ) 𝒆 𝒘 → (𝟏𝟎)
𝟐𝑾 𝟐𝒘
𝒏=−∞
𝑛
• If sample values of 𝑔 (2𝑊) are known for all values of 𝑛, then 𝐺(𝑓) of the signal is
uniquely determined by eq (10).

Reconstruction of the Signal from its Samples:

Taking IFT of eq (10), we get
∞
𝑔(𝑡) = ∫ 𝐺(𝑓)𝑒 𝑗2𝜋𝑓t ⋅ 𝑑𝑓 → (11)
−∞
Substituting eq (10) in eq (11), we get
𝑾 ∞
𝟏 𝒏 −𝒋𝝅𝒏𝒇
𝒈(𝒕) = ∫ ∑ 𝒈 ( ) 𝒆 𝒘 ⋅ 𝒆𝒋𝟐𝝅𝒇𝒕 ⋅ 𝒅𝒇
−𝒘 𝟐𝒘 𝟐𝒘
𝒏=−∞
Interchanging the order of Integration & Summation & Combining the exponential, we get
∞ 𝑤
1 𝑛 𝑛
𝑗2𝜋𝑓(𝑡− )
𝑔(𝑡) = ∑ 𝑔( )∫ 𝑒 2𝑤 ⋅ 𝑑𝑓
2𝑊 2𝑤 −𝑊
𝑛=−∞
𝑛 𝑤
∞ 𝑗2𝜋𝑓(𝑡− )
𝑛 1 𝑒 2𝑤 𝒆𝒂𝒇
𝑔(𝑡) = ∑ 𝑔 ( ) [ ] ∵ ∫ 𝒆𝒂𝒇 ⋅ 𝒅𝒇 =
2𝑤 2𝑤 𝑗2𝜋 (𝑡 − 𝑛 ) 𝒂
𝑛=−∞ 2𝑤 −𝑤
∞ 𝑛 𝑛
𝑗2𝜋𝑤(𝑡− ) −𝑗2𝜋𝑤(t− )
𝑛 1 𝑒 2𝑤 −𝑒 2𝑤
𝑔(𝑡) = ∑ 𝑔 ( ) [ 𝑛 ]
2𝑤 2𝑤 𝑗2𝜋 (t − 2𝑤 )
𝑛=−∞
𝑗𝜃
− 𝑒 −𝑗𝜃
𝑒 𝑛
sin 𝜃 = , Where 𝜃 = 2𝜋𝑤 (𝑡 − )
2𝑗 2𝑁
∞ 𝑛
𝑛 1 sin 2𝜋𝑤 (𝑡 − 2𝑤 )
𝑔(𝑡) = ∑ 𝑔 ( ) 𝑛
2𝑤 2𝑤 𝜋 (𝑡 − 2𝑤 )
𝑛=−∞
∞ 2𝜔𝑡 − 𝑛
𝑛 sin 2𝜋𝑤 ( 2𝑤 )
𝑔(𝑡) = ∑ 𝑔 ( )
2𝑤 2𝑤𝜋 (2𝜔𝑡 − 𝑛)
𝑛=−∞ 2𝜔
∞
𝑛 sin 𝜋(2𝜔𝑡 − 𝑛)
𝑔(𝑡) = ∑ 𝑔 ( )
2𝜔 𝜋(2𝜔𝑡 − 𝑛)
𝑛=−∞
∞
𝒏
𝒈(𝒕) = ∑ 𝒈 ( ) 𝐬𝐢𝐧 𝒄(𝟐𝒘𝒕 − 𝒏) → (𝟏𝟐)
𝟐𝒘
𝒏=−∞

𝐬𝐢𝐧 𝝅𝜽
𝑊𝑘𝑡 = 𝒔𝒊𝒏𝒄𝜽
𝝅𝜽

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Eq (12) is known as Interpolation formula for reconstruction of 𝑔(𝑡) from sequence of samples
𝑛
𝑔 ( ) each Sample is multiplied by a delayed 𝒔𝒊𝒏𝒄 function. All delayed 𝒔𝒊𝒏𝒄 function are added
2𝑤
to obtain 𝑔(𝑡).

Explanation in time-domain:

Reconstruction filter

If we pass Samples g(nTS ) through a filter whose impulse response is 𝒔𝒊𝒏𝒄 function then output
is the convolution of input & 𝒔𝒊𝒏𝒄 function.
The result is periodic repetition of 𝒔𝒊𝒏𝒄 pulses weighted by sample magnitude.

Frequency domain:

Fig. 2: Frequency domain representation

Sampled Signals in frequency domain is the periodic repetition of the spectrum of original
message signal at the instants ' 𝑛𝑓𝑠 '. When this signal is passed through filter whose response is
rectangle ranges between 𝑊 & 𝑊 centered at ‘0’ HZ known as LPF. The output of LPF is spectrum
component ranging between W & W centered at ‘0’ HZ. Hence it is the original message spectrum.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Case i: When 𝒇𝑺 = 𝟐 𝐖
When 𝒇𝑺 = 𝟐𝑾, information signal can be received from its samples ideally.
Practically, no filter has sharp roll-off. Therefore practically, it is NOT possible to recover the
signal 𝑔(𝑡).
𝟏 𝟏
𝒇𝑺 = 𝟐𝑾 is known as Nyquist rate & 𝑻𝑺 = 𝒇 = 𝟐𝑾 is known as Nyquist Interval.
𝑺

Case ii: When 𝒇𝒔 < 𝟐𝑾

When 𝒇𝑺 < 𝟐𝑾, there will be an overlapping of the spectrum component. Hence output of LPF
will have distortion due to unwanted frequency components. This is known as "Aliasing". This
type of Sampling is known as "Under Sampling."
Aliasing can be overcome by using LPF as an antialiasing filter before sampling which will strictly
band limit the Signal to be Sampled. Second technique of eliminating aliasing is increasing the
Sampling.

Case iii: When 𝒇𝑺 > 𝟐𝑾

When 𝒇𝑺 > 𝟐𝐖, there will be "Guard Band" between Components of the spectrum components.
Hence original Signal can be reconstructed from the samples. The 𝒇𝑺 = 𝟐. 𝟐 𝐖 is the commercially
used sampling rate & is knowm as engineers sampling rate.

Disadvantages of Ideal Sampling:

The disadvantage of ideal sampling is that due to very narrow pulses, the transmitted power is
very small & Signal to Noise ratio is very low. Thus, ideally sampled pulses may get lost in the
background noise.
Ideal Sampling is not possible to achieve practically because practically it is impossible to have
pulses of width approaching Zero.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Practical aspects of Sampling:

The sampled pulses will have finite duration rather than impulses. Amplitude of the pulses is also
finite.

The practical reconstruction filters are not Ideal filters. These Filters need a guard band or gap
between the spectrum component of Sampled Signal.

There is always problem in Selection of '𝑓'.

There are two types of practical Sampling:
1 Natural Sampling
2 Flat Top Sampling

Pulse Amplitude Modulation (PAM) OR Flat - Top Sampling OR Rectangular pulse Sampling:
As the name itself indicates after sampling, the pulses will have "Flat Top". It is very easy to obtain
flat of Sampler.

Fig. 1: A Sample & hold circuit to generate Flat-Top Samples

The sample and Hold (S/H) circuit consists of two field effect transistors (FET) switches and a
capacitor. The sampling switch is closed for a short duration by a short pulse applied to the gate
G1 of the transistor. During this period, the capacitor ‘C’ is quickly charged up to a voltage equal
to the instantaneous sample value of the incoming signal. Now, the sampling switch is opened and
the capacitor ‘C’ holds the charge. The discharge switch is then closed by a pulse applied to gate
G2 of the other transistor. Due to this, the capacitor ‘C’ is discharged to zero volts. The discharges
switch is then opened and thus capacitor have no voltage. Hence, the output of the sample and
hold circuit consists of a sequence of flat top samples.
The Top of the Samples remains constant & equal to Instantaneous value of the base band Signal
𝑔(𝑡) at the Start of sampling.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Fig. 2: Flat-Top sampling or PAM

It is observed from the Figure that only Starting edge of the pulse represents the Instantaneous
value of base band signal 𝑔(𝑡).
The sampled Signal 𝑔∂ (𝑡) is given by
𝑔∂ (𝑡) = 𝑔(𝑡) ⋅ 𝑆⸹ (𝑡)
∞

𝑔∂ (𝑡) = 𝑔(𝑡) ⋅ ∑ ∂(𝑡 − 𝑛𝑇𝑠 )

𝑛=−∞
∞

𝒈⸹ (𝒕) = ∑ 𝒈(𝒏𝑻𝒔 )𝛛(𝒕 − 𝒏𝑻𝒔 ) → (𝟏)

𝒏=−∞
Correlating 𝑔⸹ (𝑡) with the pulse ℎ(𝑡), we get
𝑆(𝑡) = 𝑔∂ (𝑡) ∗ ℎ(𝑡)
∞
𝑺(𝒕) = ∫ 𝒈𝛛 (𝝉) ⋅ 𝒉(𝒕 − 𝝉)𝒅𝝉 → (𝟐)
−∞
Substituting eq (1) in eq (2), we get
∞ ∞

𝑆(𝑡) = ∫ ∑ 𝑔(𝑛𝑇𝑠 ) ∂(𝜏 − 𝑛𝑇𝑠 )ℎ(𝑡 − 𝜏)𝑑𝜏

−∞ 𝑛=−∞

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Interchanging the order of Integration & summation, we get

∞ ∞
𝑺(𝒕) = ∑ 𝒈(𝒏𝑻𝑺 ) ∫ 𝛛(𝝉 − 𝒏𝑻𝒔 ) ⋅ 𝒉(𝒕 − 𝝉)𝒅𝝉 → (𝟑)
⏟
𝒏=−∞ −∞
from shifting property
∞
∫ 𝑥(𝑡) ⋅ ∂(𝑡 − 𝑡0 )𝑑𝑡 = 𝑥(𝑡0 )
−∞
∞
∴ ∫ ℎ(𝑡 − 𝜏) ∂(𝜏 − 𝑛𝑇𝑠 )𝑑𝜏 = ℎ(𝑡 − 𝑛𝑇𝑆 )
−∞

Shifting property in eq (3), we get

∞

𝑆(𝒕) = ∑ 𝒈(𝒏𝑻𝒔 ) ⋅ 𝒉(𝒕 − 𝒏𝑻𝒔 ) → (𝟒)

𝒏=−∞
Eq (4) represents the value of 𝑆(𝑡) interms of Sampled value 𝑔(𝑛𝑇𝑠 ) & function ℎ(t − 𝑛𝑇𝑠 ) for
flat Top sampled Signal.

𝑊𝑘𝑡
𝑆(𝑡) = 𝑔⸹ (𝑡) ∗ ℎ(𝑡)
Taking F.T on both Sides of above equation, we get
𝑆(𝑓) = 𝐺⸹ (𝑓) ⋅ 𝐻(𝑓)
Substituting 𝐺⸹ (𝑓) in above equation, we get
∞

𝑺(𝒇) = 𝒇𝑺 ∑ 𝑮(𝒇 − 𝒏𝒇𝑺 ) ⋅ 𝑯(𝒇) → (𝟓)

𝒏=−∞
Eq. (5) represents the Spectrum of Flat Top Sampled Signal.

Problem 1.
Find out the continuous time Fourier transform of rectangular pulse. Also draw the
magnitude spectrum of the output.
AKTU, 2021-22 - 10 Marks
Solution:

Fig. 1: Rectangular Pulse

The mathematical expression for the rectangular pulse depicted in figure 1 is as follows:
𝑨 for − 𝑻/𝟐 ≤ 𝒕 ≤ 𝑻/𝟐
rect (𝒕/𝑻) = {
𝟎 elsewhere

The Fourier transform is represented by the following equation

∞
𝑭{𝒙(𝒕)} = 𝑿(𝒇) = ∫ 𝒙(𝒕) ⋅ 𝒆−𝒋𝟐𝝅𝒇𝒕 𝒅𝒕 → (𝟏)
−∞
𝑇
2 𝐴 𝑇/2
𝐹{𝑥(𝑡)} = ∫ 𝐴 𝑒 −𝑗2𝜋𝑓𝑡 𝑑𝑡 = [𝑒 −𝑗2𝜋𝑓𝑡 ]−𝑇/2
−
𝑇 −𝑗2𝜋𝑓
2
𝐴
Hence, 𝐹{𝑥(𝑡)} = [𝑒 −𝑗𝜋𝑓𝑇 − 𝑒 𝑗𝜋𝑓𝑡 ]
−𝑗2𝜋𝑓
𝐴
𝐹{𝑥(𝑡)} = [𝑒 𝑗𝜋𝑓𝑇 − 𝑒 −𝑗𝜋𝑡𝑡 ]
𝑗2𝜋𝑓

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

𝐴 𝑒 𝑗𝜋𝑓𝑇 − 𝑒 −𝑗𝜋𝑓𝑇
𝐹{𝑥(𝑡)} = [ ]
𝜋𝑓 2𝑗
We know that
𝐞𝒋𝜽 − 𝐞−𝐣𝜽
𝐬𝐢𝐧 𝜽 =
𝟐𝐣
𝐀
𝐅{𝐱(𝐭)} = [𝐬𝐢 𝐧(𝝅𝐟)] → (𝟐)
𝝅𝐟
𝑀𝑢𝑙𝑡𝑖𝑝𝑙𝑦𝑖𝑛𝑔 & dividing the RHS of Eq (2) by ' 𝑇 ', we get
sin(𝜋𝑓𝑇)
𝐹[𝑥(𝑡)] = AT
𝜋𝑓𝑇
Wkt
𝐬𝐢𝐧(𝝅𝒇𝑻)
𝐬𝐢𝐧𝐜(𝒇𝑻) =
𝝅𝒇𝑻
𝑭[𝒙(𝒕)] = At 𝐬𝐢𝐧 𝐜(𝒇𝑻) → (𝟑)
Therefore, the rectangular pulse transforms into a sinc function.
We know that
𝑠𝑖𝑛𝑐 (0) = 1
Therefore, 𝐴𝑇 𝑠𝑖𝑛𝑐 (0) = AT (1) = AT
The sinc function will have zero value for the following values
𝐬𝐢𝐧𝐜(𝐟𝐓) = 𝟎 for 𝐟𝐓 = ±𝟏, ±𝟐, ±𝟑, … …
1 2 3
i.e. for f = ± T , ± T , ± T … …..

Fig. 2: Amplitude and Phase Spectrum of rectangular pulse

The amplitude and phase spectrum of the rectangular function is as shown in Fig 2. By introducing
a phase shift of ±180 degrees in the phase spectrum, the negative amplitude of the amplitude
spectrum |X(f)| can be transformed into a positive value. This phenomenon is illustrated in figure
2. To ensure symmetry in the phase spectrum, a negative phase shift is applied for positive
frequencies, while a positive phase shift is applied for negative frequencies.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Problem 2.
Obtain the expression for Fourier transform of Sampling function 𝒉(𝒕) used for flat top
Sampling.
Solution:

𝑻
𝑯(𝒇) = ∫ 𝒉(𝒕)𝒆−𝒋𝟐𝝅𝒇𝒕 ⋅ 𝒅𝒕 → (𝟏)
𝟎
𝑇
𝐻(𝑓) = ∫ 1 ⋅ 𝑒 −𝑗2𝜋𝑓𝑡 ⋅ 𝑑𝑡
0
−𝑗2𝜋𝑓𝑡 𝑇
𝑒
= ]
−𝑗2𝜋𝑓 0
1
= [𝑒 −𝑗2𝜋𝑓𝑇 − 𝑒 0 ]
−𝑗2𝜋𝑓

1
𝐻(𝑓) = [𝑒 −𝑗2𝜋𝑓𝑇 − 𝑒 0 ]
−𝑗2𝜋𝑓
1
𝐻(𝑓) = [𝑒 −𝑗2𝜋𝑓𝑇 − 1]
−𝑗2𝜋𝑓
1
𝐻(𝑓) =+ [1 − 𝑒 −𝑗2𝜋𝑓𝑇 ⋅]
𝑗2𝜋𝑓
𝑗𝑠𝜋𝑓𝑇 𝑗2𝜋𝑓𝑇 𝑗2𝜋𝑓𝑇 1 − 𝑒 −𝜃 𝑒 −𝜃/2 [𝑒 𝜃/2 − 𝑒 −𝜃/2 ]
𝑒− 2 [𝑒 2 − 𝑒− 2 ] =
𝐻(𝑓) = 𝜃 𝜃
2𝑗𝜋𝑓 𝑒 −𝜃/2+𝜃/2 − 𝑒 −𝜃/2−𝜃/2
=
𝜃
𝑒 −𝑗𝜋𝑓𝑇 [𝑒 𝑗𝜋𝑓𝑇 − 𝑒 −𝑗𝜋𝑓𝑇 ]
𝐻(𝑓) = 𝟏 − 𝒆−𝜽 𝒆𝟎 − 𝒆−𝜽
2𝑗𝜋𝑓 =
𝜽 𝜽

𝒔𝒊𝒏 𝝅𝒇𝑻
𝑯(𝒇) = 𝒆−𝒋𝝅𝒇𝑻 ⟶ (2)
𝝅𝒇

𝑀𝑢𝑙𝑡𝑖𝑝𝑙𝑦𝑖𝑛𝑔 & 𝑑𝑖𝑣𝑖𝑑𝑖𝑛𝑔 Eq. (2) by 𝑻, we get

𝒆𝒋𝜽 − 𝒆−𝒋𝜽
−𝒋𝝅𝒇𝑻
𝒔𝒊𝒏 𝝅𝒇𝑻 ∵ 𝐬𝐢𝐧 𝜽 =
𝑯(𝒇) = 𝑻𝒆 ⋅ 𝟐𝒋
𝝅𝒇 ⋅ 𝑻

𝐻(𝑓) = 𝑇𝑒 −𝑗𝜋𝑓𝑇 𝑠𝑖𝑛𝑐 𝑓𝑇

𝐬𝐢𝐧𝝅𝜽
∵ 𝐬𝐢𝐧𝒄 𝜽 =
𝑯(𝒇) = 𝑻 𝐬𝐢𝐧𝒄(𝒇𝑻)𝒆−𝒋𝝅𝒇𝑻 𝝅𝜽

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Explain aperture effect with the help of Spectral diagrams. Bring out the differences
between aperture effect & aliasing effect.
ALIASING EFFECT:
Aliasing effect is due to wrong Choice of Sampling frequency 𝑓𝑠 . When 𝑓𝑠 < 2W, there will be
overlapping of spectrum component. Hence output of LPF will have distortion due to unwanted
frequency Component. This is known of "Aliasing" effect.

Fig. 2: Aliasing effect

Aperture effects with Spectral diagrams:

The high frequency roll-off of 𝐻(𝐹) acts as LPF & attenuates upper portion of the message
spectrum. This effect is known as "APERTURE EFFECT".

Fig. 3: Aperture effect

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

FORMULAE

1 Nyquist rate 𝑓𝑠 = 2𝑤
1
2 Sampling period 𝑇𝑠 =
𝑓𝑠
𝜔
3 𝜔 = 2𝜋𝑓𝑚 , 𝑓𝑚 = 2𝜋
1
4 cos 𝐴 ⋅ cos 𝐵 = 2 [cos(𝐴 − 𝐵) + cos(𝐴 + 𝐵)]
sin 𝜋𝜃
5 sinc 𝜃 = 𝜋𝜃
sin2 𝜋𝜃
6 sin2 𝜃 = (𝜋𝜃)2
1−cos 2𝜃
7 sin2 𝜃 = 2
𝐹𝑇 𝐴𝐶
8 𝐴𝑐 cos 2𝜋𝑅𝐶 𝑡 ⟶ [∂(𝑓 − 𝑓𝑐 ) + ∂(𝑓 + 𝑓𝑐 )]
2
𝐹𝑇 𝐴
9 𝐴𝑐 sin 2𝜋𝑓𝑐 t ⟶ 2𝑗𝑐 [∂(𝑓 − 𝑓𝑐 ) + ∂(𝑓 + 𝑓𝑐 )]

10 𝐻(𝑓) = T 𝑆inc(𝑓𝑇)𝑒 −𝑗𝜋𝑓𝑇

11 In phase & Quadrature component
𝑔(𝑡) = 𝑔𝐼 (𝑡)cos(2𝜋𝑓𝑐 𝑡) − 𝑔𝑄 (𝑡)sin(2𝜋𝑓𝑐 𝑡)
𝐹𝑇 𝐴 𝑓
12 𝐴 sinc 2𝑤t ⟶ rect [ ]
2𝑊 2𝑊
∞ 2 ∞
13 𝐸 = ∫−∞ [|g(𝑡)|] 𝑑𝑡 = ∫−∞ 𝑔(𝑡) ⋅ 𝑔∗ (𝑡) ⋅ 𝑑𝑡
14) 𝐺⸹ (𝑓) = 𝑓𝑠 ∑∞
𝑛=−∞ 𝐺(𝑓 − 𝑛𝑓𝑠 )

Problem 3.
An analog signal is expressed by the equation, calculate the Nyquist rate & Nyquist Interval
for this Signal.
𝒙(𝒕) = 𝟑𝐜𝐨𝐬 𝟓𝟎𝝅𝒕 + 𝟏𝟎𝐬𝐢𝐧 𝟑𝟎𝟎𝝅𝒕 + 𝐜𝐨𝐬 𝟏𝟎𝟎𝝅𝒕.

Solution:
𝒙(𝒕) = 𝟑 𝐜𝐨𝐬 𝟓𝟎𝝅𝒕 + 𝟏𝟎 𝐬𝐢𝐧 𝟑𝟎𝟎𝝅𝒕 + 𝐜𝐨𝐬 𝟏𝟎𝟎𝝅𝒕 → (𝟏)
equation (1) is in the form
𝑥(𝑡) = 3 cos 𝜔1 𝑡 + 10 sin 𝜔2 𝑡 + cos 𝜔3 𝑡 → (2)
comparing Eq. (1) & (2), we get

𝜔1 = 50𝜋 𝜔2 = 300𝜋 𝜔3 = 100𝜋

2π 𝑓1 = 50𝜋 2𝜋𝑓2 = 300𝜋 2𝜋𝑓3 = 100𝜋
𝒇𝟏 = 𝟐𝟓 𝐇𝐳 𝒇𝟐 = 𝟏𝟓𝟎 𝐇𝐳 𝒇𝟑 = 𝟓𝟎 𝐇𝐳

• 𝒇𝒎 is given by
𝑓𝑚 = 𝑚𝑎𝑥(𝑓1 , 𝑓2 , 𝑓3 ) = 𝑚𝑎𝑥 (25, 150, 50)
𝒇𝒎 = 𝟏𝟓𝟎 𝑯𝐳

• Nyquist rate ' 𝑓𝑠′ = 2𝑓𝑚 = 2 × 150 Hz

𝒇𝑺 = 𝟑𝟎𝟎 𝑯𝒛

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

1 1
• Nyquist Interval ' 𝑇𝑠′ = =
𝑓𝑠 300Hz
𝑻𝒔 = 𝟑. 𝟑 msec

Problem 4.
Find the Nyquist rate & Nyquist Interval for the Signal
𝟏
𝒙(𝒕) = 𝐜𝐨𝐬(𝟒𝟎𝟎𝟎𝝅𝒕)𝐜𝐨𝐬(𝟏𝟎𝟎𝟎𝝅𝒕)
𝟐𝝅
Given
1
𝑥(𝑡) = cos(4000𝜋𝑡)cos(1000𝜋𝑡)
2𝜋
Solution:

𝟏
Wkt 𝐜𝐨𝐬 𝑨 ⋅ 𝐜𝐨𝐬 𝑩 = 𝟐 [𝐜𝐨𝐬(𝑨 − 𝑩) + 𝐜𝐨𝐬(𝑨 + 𝑩)]
1
𝑥(𝑡) = [cos(4000𝜋 − 1000𝜋)𝑡 + cos(4000𝜋 + 1000𝜋)𝑡]
4𝜋
1
𝑥(𝑡) = [cos(3000𝜋𝑡) + cos(5000𝜋𝑡)]
4𝜋
Eq. (2) is in the form of
1
𝑥(𝑡) = [cos 𝜔1 𝑡 + cos 𝜔2 𝑡]
4𝜋

𝜔1 = 3000𝜋 𝜔2 = 5000𝜋
2π 𝑓1 = 3000𝜋 2𝜋𝑓2 = 5000𝜋
𝒇𝟏 = 𝟏𝟓𝟎𝟎 𝐇𝐳 𝒇𝟐 = 𝟐𝟓𝟎𝟎 𝐇𝐳
• 𝒇𝒎 is given by

𝑓𝑚 = 𝑚𝑎𝑥(𝑓1 , 𝑓2 ) = 𝑚𝑎𝑥 (1500, 2500)

𝒇𝒎 = 𝟐𝟓𝟎𝟎 𝑯𝐳

Nyquist rate ' 𝑓𝑠 = 2 × 𝑓𝑚 = 2 × 2500 Hz

𝒇𝒔 = 𝟓𝟎𝟎𝟎𝐇𝐳
Nyquist Interval 𝑇𝑠 = 1/𝑓𝑠 = 1/5000 Hz
𝑻𝑺 = 𝟎. 𝟐𝐦𝐬𝐞𝐜

Problem 5.
A Signal 𝑔(𝑡) = 2cos 400𝜋𝑡 + 6cos 640𝜋𝑡 is ideally sampled at 𝑓𝑠 = 500𝐻𝑧. If the Sampled Signal
is passed through an ideal low pass filter with cut-off frequency 𝑓𝑐 = 400 Hz. Find
i) 𝐺(𝑓) & Sketch its Spectrum
ii) Sampled Signal 𝐺∂ (𝑓) & Sketch its Spectrum.
iii) The components that will appear at the filter output

Solution:
𝒈(𝒕) = 𝟐 𝐜𝐨𝐬(𝟒𝟎𝟎𝝅𝒕) + 𝟔 𝐜𝐨𝐬(𝟔𝟒𝟎𝝅𝒕) → (1)

Eq. (1) is in the form of

𝑔(𝑡) = 2 cos(𝜔1 𝑡) + 6 cos(𝜔2 𝑡)

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

𝜔1 = 400𝜋 𝜔2 = 640𝜋
2π 𝑓1 = 400𝜋 2𝜋𝑓2 = 640𝜋
𝒇𝟏 = 𝟐𝟎𝟎 𝐇𝐳 𝒇𝟐 = 𝟑𝟐𝟎 𝐇𝐳

𝑔(𝑡) = 2cos 2𝜋𝑓1 𝑡 + 6cos 2𝜋𝑓2 𝑡

g(t)=2 cos 2𝝅(200) t+6 cos 2 𝝅 (320) t → (2)
Taking 𝐹𝑇 on both Sides of eq (2), we get
2 6
𝐺(𝑓) = [∂(𝑓 − 200) + ∂(𝑓 + 200)] + [∂(𝑓 − 320) + ∂(𝑓 + 320)]
2 2
𝑮(𝒇) = [𝛛(𝒇 − 𝟐𝟎𝟎) + 𝛛(𝒇 + 𝟐𝟎𝟎)] + 𝟑[𝛛(𝒇 − 𝟑𝟐𝟎) + 𝛛(𝒇 + 𝟑𝟐𝟎)] → (𝟑)

Using Eq (3), the Spectrum of the Signal 𝑔(𝑡) is drawn & it appears as Shown in fig (1).

ii) Wkt
∞

𝑮⸹ (𝒇) = 𝒇𝒔 ∑ 𝑮(𝒇 − 𝒏𝒇𝒔 )

𝒏=−∞
Giveen 𝒇𝑺 = 𝟓𝟎𝟎𝐇𝐳
∞

= 𝑓𝑆 ∑ 𝐺(𝑓 − 500𝑛)
𝑛=−∞
∞

𝑮⸹ (𝒇) = 𝒇𝑺 ∑ [𝛛(𝒇 − 𝟓𝟎𝟎𝒏 − 𝟐𝟎𝟎) + 𝛛(𝒇 − 𝟓𝟎𝟎𝒏 + 𝟐𝟎𝟎)]

𝒏=−∞
∞

+𝟑𝒇𝒔 ∑ [𝛛(𝒇 − 𝟓𝟎𝟎𝒏 − 𝟑𝟐𝟎) + 𝛛(𝒇 − 𝟓𝟎𝟎𝒏 + 𝟑𝟐𝟎)] → (𝟒)

𝒏=−∞

• The Spectrum of the Sampled Signal g(𝑡) is drawn using eq (4)

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

iii) Spectrum of Ideal LPF:

The components that are appeared at the filter alp are 180Hz, 𝟐𝟎𝟎 𝐇𝐳, 𝟑𝟎𝟎 𝐇𝐳, 𝟑𝟐𝟎 𝐇𝐳.

Problem 6.
The Signal 𝑔(𝑡) = 10cos 20𝜋t. cos 200𝜋𝑡 is Sampled at rate of 250 Samples/sec.
i. Determine the Spectrum of resulting Sampled Signal.
ii. Specify the cut-off frequency of the Ideal reconstruction filter so as to recover 𝑔(𝑡) from
its Sampled version.
iii. What is the Nyquist rate for 𝑔(𝑡) ?

Solution:
𝒈(𝒕) = 𝟏𝟎 𝐜𝐨𝐬 𝟐𝟎𝝅𝒕 . 𝐜𝐨𝐬 𝟐𝟎𝟎𝝅𝒕 → (𝟏)
𝟏
Wkt 𝐜𝐨𝐬 𝑨 ⋅ 𝐜𝐨𝐬 𝑩 = [𝐜𝐨𝐬(𝑨 − 𝑩) + 𝐜𝐨𝐬(𝑨 + 𝑩)]
𝟐
10
𝑔(𝑡) = [cos(200 − 20)𝜋𝑡 + cos(200 + 20)𝜋𝑡]]
2
𝑔(𝑡) = 5[cos 180𝜋𝑡 + cos 220𝜋𝑡]
𝒈(𝒕) = 𝟓 𝐜𝐨𝐬 𝟏𝟖𝟎𝝅𝒕 + 𝟓 𝐜𝐨𝐬 𝟐𝟐𝟎𝝅𝒕 → (𝟐)
From Eq (2)

𝜔1 = 180𝜋 𝜔2 = 220𝜋
2π 𝑓1 = 180𝜋 2𝜋𝑓2 = 220𝜋
𝒇𝟏 = 𝟗𝟎 𝐇𝐳 𝒇𝟐 = 𝟏𝟏𝟎 𝐇𝐳

• 𝒇𝒎 is given by

𝑓𝑚 = 𝑚𝑎𝑥(𝑓1 , 𝑓2 ) = 𝑚𝑎𝑥 (90, 110)

𝒇𝒎 = 𝟏𝟏𝟎 𝑯𝐳

𝒈(𝒕) = 𝟓 𝐜𝐨𝐬 𝟐𝝅(𝟗𝟎) 𝒕 + 𝟓 𝐜𝐨𝐬 𝟐𝝅(𝟏𝟏𝟎) 𝒕 → (3)

Taking FT on both Sides of Eq. (3), we get

5 5
𝐺(𝑓) = [∂(𝑓 − 90) + ∂(𝑓 + 90)] + [∂(𝑓 − 110) + ∂(𝑓 + 110)]
2 2
𝑮(𝒇) = 𝟐. 𝟓[𝛛(𝒇 − 𝟗𝟎) + 𝛛(𝒇 + 𝟗𝟎)] + 𝟐. 𝟓[𝛛(𝒇 − 𝟏𝟏𝟎) + 𝛛(𝒇 + 𝟏𝟏𝟎)] → (𝟒)

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

The spectrum of the signal 𝑔(𝑡) is drawn using Eq (4) is shown in figure below:

i) 𝑊𝑘𝑡
∞

𝑮⸹ (𝒇) = 𝒇𝒔 ∑ 𝑮(𝒇 − 𝒏𝒇𝒔 )

𝒏=−∞
Given: 𝒇𝑺 = 𝟐𝟓𝟎
∞

𝑮𝛛 (𝒇) = 𝟐. 𝟓 𝒇𝑺 ∑ [𝛛(𝒇 − 𝟐𝟓𝟎𝒏 − 𝟗𝟎) + 𝛛(𝒇 − 𝟐𝟓𝟎𝒏 + 𝟗𝟎)]

𝒏=−∞
∞

+𝟐. 𝟓𝒑𝑺 ∑ [𝛛(𝒇 − 𝟐𝟓𝟎𝒏 − 𝟏𝟏𝟎) + 𝛛(𝒇 − 𝟐𝟓𝟎𝒏 + 𝟏𝟏𝟎)]

𝒏=−∞

The cut-off frequencies of the Ideal LPF Should be more than 𝟏𝟏𝟎 𝐇𝐳 & less than 𝟏𝟒𝟎 𝐇𝐳 for
recovering 𝑔(𝑡) from 𝑔⸹ (𝑡).

iii) Nyquist rate for 𝑔(𝑡) :

𝑓𝑠 = 2𝑓𝑚 = 2 × 110 Hz
𝒇𝒔 = 𝟐𝟐𝟎 𝐇𝐳

Problem 7.
Consider the Signal 𝑔(𝑡) = 𝐴sin(2𝜋𝑓0 𝑡). Plot the spectrum of the discrete time signal 𝑔⸹ (𝑡)
derived by sampling 𝑔(𝑡) at the time 𝑡𝑛 = 𝑛 ∣ 𝑓𝑠 ⋅ Whore 𝑛 = 0, ±1, ±2, and ….
Determine
i. 𝑓𝑠 = 𝑓0
ii. 𝑓𝑠 = 2𝑓0
iii. 𝑓𝑠 = 3𝑓0

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Given:
𝒈(𝒕) = 𝑨 𝒔𝒊𝒏(𝟐𝝅𝒇𝟎 𝒕) → (𝟏)
Solution:
Taking 𝐹𝑇 on both Sides of Eq. (1), we get
𝑨
𝑮(𝒇) = [𝛛(𝒇 − 𝒇𝟎 ) + 𝛛(𝒇 + 𝒇𝟎 )] → (𝟐)
𝟐𝒋

Fig (1): Spectrum of message Signal 𝒈(𝒕).

Wkt
∞

𝑮⸹ (𝒇) = 𝒇𝑺 ∑ 𝑮(𝒇 − 𝒏𝒇𝑺 ) → (𝟑)

𝒏=−∞
i) 𝒇𝑺 = 𝒇𝟎
𝐴𝑓0 ∞
𝐺⸹ (𝑓) = ∑ [∂(𝑓 − 𝑓0 − 𝑛𝑓0 ) + ∂(𝑓 + 𝑓0 − 𝑛𝑓0 )]
2𝑗 𝑛=−∞

ii) 𝒇𝑺 = 𝟐𝒇𝟎
∞
𝐴
𝐺0 (𝑓) = 2𝑓0 ∑ [∂(𝑓 − 𝑓0 − 𝑛2𝑓0 ) + ∂(𝑓 + 𝑓0 − 𝑛2𝑓0 )]
2𝑗
𝑛=−∞

iii) 𝒇𝒔 = 𝟑𝒇𝟎
∞
3𝐴𝑓0
𝐺𝛾 (𝑓) = ∑ [∂(𝑓 − 𝑓0 − 𝑛3𝑓0 ) + ∂(𝑓 + 𝑓0 − 𝑛3𝑓0 )]
2𝑗
𝑛=−∞

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Problem 8.
Consider the Signal 𝑔(𝑡) = 𝐴cos(2𝜋𝑓0 𝑡). Plot the spectrum of the discrete time signal 𝑔⸹ (𝑡)
derived by sampling 𝑔(𝑡) at the time 𝑡𝑛 = 𝑛 ∣ 𝑓𝑠 ⋅ Whore 𝑛 = 0, ±1, ±2, and ⋯
Determine
i. 𝑓𝑠 = 𝑓0
ii. 𝑓𝑠 = 2𝑓0
iii. 𝑓𝑠 = 3𝑓0

Given:
𝒈(𝒕) = 𝑨 𝐜𝐨𝐬(𝟐𝝅𝒇𝟎 𝒕) → (𝟏)
Solution:
Taking 𝐹𝑇 on both Sides of Eq. (1), we get
𝑨
𝑮(𝒇) = [𝛛(𝒇 − 𝒇𝟎 ) + 𝛛(𝒇 + 𝒇𝟎 )] → (𝟐)
𝟐

Fig (1): Spectrum of message Signal 𝒈(𝒕).

Wkt
∞

𝑮⸹ (𝒇) = 𝒇𝑺 ∑ 𝑮(𝒇 − 𝒏𝒇𝑺 ) → (𝟑)

𝒏=−∞
i) 𝒇𝑺 = 𝒇𝟎
𝐴𝑓0 ∞
𝐺⸹ (𝑓) = ∑ [∂(𝑓 − 𝑓0 − 𝑛𝑓0 ) + ∂(𝑓 + 𝑓0 − 𝑛𝑓0 )]
2 𝑛=−∞

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

ii) 𝒇𝑺 = 𝟐𝒇𝟎
∞
𝐴
𝐺0 (𝑓) = 2𝑓0 ∑ [∂(𝑓 − 𝑓0 − 𝑛2𝑓0 ) + ∂(𝑓 + 𝑓0 − 𝑛2𝑓0 )]
2
𝑛=−∞

iii) 𝒇𝒔 = 𝟑𝒇𝟎
∞
3𝐴𝑓0
𝐺𝛾 (𝑓) = ∑ [∂(𝑓 − 𝑓0 − 𝑛3𝑓0 ) + ∂(𝑓 + 𝑓0 − 𝑛3𝑓0 )]
2
𝑛=−∞

Problem 9.
The spectrum of band pass signal 𝑔(𝑡) has a bandwidth of 0.8kHz centered around ±10KHz.
White the equation for 𝑔(𝑡) inters of quadrature components. Find the Nyquist rate & Nyquist
interval.

Given: 𝑓𝐶 = ±10kHz, 𝐵𝑊 = 0.8kHZ.

Solution:
Spectrum of 𝐺(𝑓)

𝑔(𝑡) can be expressed interns of In-phase & Quadrature phase Components.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

∴ 𝑔(𝑡) = 𝑔𝐼 (𝑡)cos(2𝜋𝑓𝑐 𝑡) − 𝑔𝑄 (𝑡)sin(2𝜋𝑓𝑐 𝑡)

𝑔(𝑡) = 𝑔𝐼 (𝑡)cos(2𝜋 × 10 × 103 𝑡) − 𝑔𝑄 (𝑡)sin(2𝜋 × 10 × 103 𝑡)
𝐵𝑊 0.8KHz
• 𝑾= = = 0.4KHz
2 2
∴ Nyquist rate 𝑓𝑆 = 2𝑊 = 2 × 0.4kHz
𝒇𝒔 = 𝟎. 𝟖𝐊𝐇𝐳

Problem 10.
The spectrum of band pass signal 𝑔(𝑡) has a bandwidth of 0.6kHz centered around ±12KHz.
White the equation for 𝑔(𝑡) inters of quadrature components. Find the Nyquist rate & Nyquist
interval.

Given: 𝑓𝐶 = ±12kHz, 𝐵𝑊 = 0.6kHZ.

Solution:

Spectrum of 𝐺(𝑓)

𝑔(𝑡) can be expressed interns of In-phase & Quadrature phase Components.

∴ 𝑔(𝑡) = 𝑔𝐼 (𝑡)cos(2𝜋𝑓𝑐 𝑡) − 𝑔𝑄 (𝑡)sin(2𝜋𝑓𝑐 𝑡)

𝑔(𝑡) = 𝑔𝐼 (𝑡)cos(2𝜋 × 12 × 103 𝑡) − 𝑔𝑄 (𝑡)sin(2𝜋 × 12 × 103 𝑡)

𝐵𝑊 0.6KHz
• 𝑾 = 2 = 2 = 0.3KHz
∴ Nyquist rate 𝑓𝑆 = 2𝑊 = 2 × 0.3kHz
𝒇𝒔 = 𝟎. 𝟔𝐊𝐇𝐳

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

PULSE CODE MODULATION

PCM is an analog-to-digital converter where the Information Contained in the Instantaneous
samples of an analog signal is represented by digital Codes in a Serial bitstream manner.

Fig (1): Basic elements of a PCM system. (a) Transmitter. (b) Transmission path. (c)
Receiver.
The block diagram of a PCM System is Shown in Fig (1). It consists of Transmitter, Regenerative
repeater & Receiver.

PCM Transmitter:
The low-pass filter, also known as the pre-alias filter, is implemented before the sampler. Its
purpose is to limit the frequency content of the message signal to a maximum value denoted as
'W' Hz. By doing so, any frequencies above 'W' Hz are attenuated or removed from the signal. This
band limiting helps prevent aliasing, which can distort the reconstructed signal during the
sampling process.

Sampler: The incoming message Signal is sampled with a train of Narrow rectangular pulses. The
Sampling rate ' fs ' is selected above Nyquist rate to avoid aliasing i.e. 𝑓𝑆 ⩾ 2𝑊.

Quantization: The sampled Signal is fed to the quantizer. The quantizer approximates each Input
Signal level to the nearest Prefixed level. The output of the quantizer is a discrete-time discrete
valued Signal known as a "Quantized signal".

Encoding: The quantized Samples are then encoded in the encoder. The process of encoding
involves allocating Some digital code to each level. These coded levels are then emitted as a
bitstream of data i.e., 0's & 1's. The encoder output Consists of pulses depending on the Code
Combination.

Regenerative repeater:
The PCM Signal is reconstructed by means of a regenerative repeater located at Sufficiently closed
Spacing along the transmission path. The regenerative networks are used at intermediate points
between the transmitter & receiver in order to Boost up the pulse amplitude.
PCM receiver:
Decoder: The first operation in the receiver is to generate the received pulses. The decoder
converts binary Coded signals to approximated pules of discrete magnitude.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Reconstruction filter: The final operation in the receiver is to recover the original analog Signal.
This is done by passing the decoder output through an LPF. The output of LPF is an analog Signal.

Advantage: -
1 Relatively inexpensive digital circuitry is involved in PCM.
2 PCM Signals can be multiplexed & transmitted over a common high-Speed Communication
link.
3 In long-distance transmission, clean waveforms can be regenerated using repeaters.
4 The noise performance of a digital System is Superior to that of an analog system.

QUANTIZATION NOISE AND SIGNAL TO NOISE RATIO IN PCM SYSTEM:

1 Derive the expression for Signal to quantization noise ratio for PCM system that employs
linear quantization technique?
2 A PCM System uses a uniform quantizer followed by a '𝑣' bit encoder. Show that rms Signal
to quantization noise ratio is approximately given by (1.8 + 6𝑣) 𝑑𝐵.
3 Derive an expression for maximum Signal to quantization noise ratio for PCM System that
employs linear quantization techniques. What will be the (SNR)dB if the destination power
& Signal amplitude are Normalized {(SNR )𝑑𝐵 = 4 ⋅ 8 + 6 N}.
4 obtain an expression for the Signal to quantization noise power Ratio in the case of PCM.
Assume that the amplitude of Signal is uniformly distributed.

Solution:
• Let the random variable ' 𝑄 ' denotes the quantization error & ' 𝑞 ' its sample value.
• Let us assume that the quantization error '𝑄' is uniformly distributed over a Single
Quantize Interval '△'.

Wkt PDF is
given by
1
𝑓𝑋 (𝑥) = {𝑏 − 𝑎 𝑎⩽𝑥⩽𝑏
0 otherwise
∴ Probability density function (PDF) of Quantization error ' 𝑄
' is then 1 1
∴ 𝑓𝑄 (𝑞) = =
Δ/2 − (−Δ/2) 2 Δ/2
1/Δ for − Δ/2 ⩽ 𝑞 ⩽ Δ/2
𝑓𝑄 (𝑞) = { 𝒇𝑸 (𝒒) = 𝟏/𝚫
0 otherwise

• The mean quantization error 𝝁 = 𝟎 W.K.T mean

• The variance of quantization error is
Δ/2 𝑎 + 𝑏 −Δ/2 + Δ/2
𝜎𝑄2 =∫ (𝑞 − 𝜇)2 ⋅ 𝑓𝑄 (𝑞) ⋅ 𝑑𝑞 𝜇= =
2 2
−Δ/2
𝝁=𝟎
Δ/2
1
𝜎𝑄2 =∫ (𝑞 − 0)2 ⋅ 𝑑𝑞
−Δ/2 Δ

Where mean 𝝁 = 𝟎

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Δ/2 Δ/2
12
1 Δ/2 2 1 𝑞3
=∫ 𝑞 ⋅ 𝑑𝑞 = ∫ 𝑞 ⋅ 𝑑𝑞 = [ ]
−Δ/2 Δ Δ −Δ/2 Δ 3 −Δ/2
1 3 Δ/2 1 Δ 3 −Δ 3 1 Δ3 −Δ3
= [𝑞 ]−Δ/2 = 3Δ [(2 ) − ( 2 ) ] = 3Δ [ 8 − 8 ]
3Δ
2
1 3 1 Δ3 1 Δ2
= [Δ /8 + Δ3 /8] = 2( ) = ⋅
3Δ 3Δ 8 3 4
𝟐
𝚫
𝝈𝟐𝟐 = ⟶ (1)
𝟏𝟐

• Eq (1) is known as "Mean Squared quantization error" or Normalized Noise power of

Quantization error interns of power.
• Let us Consider '𝑁' bits to represent ' 𝐿 ' quantized level, then 𝑳 = 𝟐𝑵
∴ Step Size

2𝑥max
Δ=
𝐿
𝟐𝒙𝐦𝐚𝐱
𝚫= → (2)
𝟐𝑵 𝑥max − (−𝑥max )
Δ =
Substituting Eq. (2) in eq (1), we get 𝐿
2𝑥max
Δ =
(2𝑥max /2𝑁 )2 2𝑁
𝜎𝑄2 =
12
2 2
4𝑥max 4𝑥max 1
𝜎𝑄2 = 2𝑁 = 2𝑁 ×
2 /12 2 12
𝟏
𝝈𝟐𝑸 = 𝒙𝟐𝐦𝐚𝐱 ⋅ 𝟐−𝟐𝑵 ⟶ (𝟑)
𝟑
• Let 'P' denotes the average power of the message signal 𝑥(𝑡), then the input SNR of a
uniform quantized is
𝑆 Signal power 𝑃
= = 2
𝑁 Noise power 𝜎𝑄
𝑃 3𝑃
= = 2 22𝑁
1 2
𝑥 2−2𝑁 𝑥max
3 max
𝟑𝑷
(𝑺𝑵𝑹)𝟎 = 𝟐 ⋅ 𝟐𝟐𝑵 ⟶ (4)
𝒙𝒎𝒂𝒙

• For Normalized input voltage 𝑥max = 1 & power 𝑝 ⩽ 1.

3(1) 2𝑁
∴ SNR = ⋅2
(1)2
SNR = 3 ⋅ 22𝑁

(𝑆𝑁𝑅)𝑑𝐵 = 10 𝑙𝑜𝑔10 (3.22𝑁 )

= 10 𝑙𝑜𝑔10 (3) + 10 𝑙𝑜𝑔10 (22𝑁 )
= 4.8 + 20𝑁 𝑙𝑜𝑔10 (2)
(𝑺𝑵𝑹)𝒅𝑩 = 𝟒. 𝟖 + 𝟔𝑵 → (5)

Eq (5) is the Normalized Signal to quantization noise ratio in 𝑑𝐵 for any message Signal.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

For Sinusoidal message Signal

Let
𝑥(𝑡) = 𝐴𝑚 cos 2𝜋𝑓𝑚 𝑡
𝒙max = 𝑨𝒎
∴ The power of this Signal is

𝑽𝟐 𝐴𝑚
𝑷= 𝑉=
𝑹 √2

When 𝑅 = 1, the power ' 𝑃 ' is normalized

2
(𝐴𝑚 /√2) 𝐴2𝑚
𝑃= =
𝑅 2×1
𝑨𝟐𝒎
𝑷=
𝟐
Wkt
3𝑃
(𝑆𝑁𝑅)0 = 2 ⋅ 22𝑁 from eq (4)
𝑥max
3(𝐴2𝑚 /2) 2𝑁
= ⋅2
𝐴2𝑚
𝟑
(𝑺𝑵𝑹)𝟎 = ⋅ 𝟐𝟐𝑵
𝟐
3
( SNR )𝑑𝐵 = 10𝑙𝑜𝑔10 ( ⋅ 22𝑁 )
2
= 10𝑙𝑜𝑔10 (3/2) + 10𝑙𝑜𝑔10 (22𝑁 )
= 1.76 + 20𝑁𝑙𝑜𝑔10 (2)
(𝑺𝑵𝑹)𝒅𝑩 = 𝟏. 𝟕𝟔 + 𝟔. 𝟎𝟐 𝑵

• Eq (6) is known as "6 dB rule" for uniform quantization. This is because each additional
bit of quantization level increases the Signal to Noise Ratio by 6 dB.

DIFFERENTIAL PULSE CODE MODULATION (DPCM):

• In PCM System, each sample of waveform is encoded independently of their Samples.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

• The Samples of Signals are highly correlated this is because

i. Any signal does not change fast.
ii. We are taking the Samples above Nyquist rate i.e., 𝑓𝑆 > 2𝑊.
Thus, the Signal does not change rapidly from one sample to the next Sample.
• When these highly correlated Samples are encoded, the resulting encodes Signal Contains
Redundant information.
• The redundancy can be eliminated by using DPCM. The Fig. shows the Continuous-time Signal
𝑥(t) Sampled using flat-top Sampling. This Signal is sampled at the instants 𝑇𝑠 , 2𝑇𝑠 , 3𝑇𝑠 , ⋯ ⋅
, 𝑛𝑇𝑠 . The Sampling frequency is selected higher than the Nyquist rate & encoded using a 3-bit
encoder. The Samples quantized to the nearest level are shown in Fig (1).
• We can see that Samples taken at 4𝑇𝑠 , 5𝑇𝑠 , 6𝑇𝑠 are encoded to same value "110". One Sample
can carry this information. But three Samples carrying the same information means it is
redundant.
• If this redundancy is reduced, then the overall bitrate will decrease & the number of bits
required to transmit one sample is also reduced. This type of digital pule modulation Scheme
is known as DPCM.
• When an analog signal is sampled at a rate slightly higher than the Nyquist rate, resulting in
highly correlated samples, i.e., the signal does not change rapidly from one Sample to the next.
• If these highly correlated Samples are encoded, the resulting encoded Signal contains
redundant information.
• If we remove this redundancy before encoding, efficiency of the coded Signal can be increased.
• DPCM works on the principle of prediction. The value of the present sample is predicted from
the past Samples. The prediction may not be exact, but it is very close to the Sample value.

DPCM Transmitter:

Fig. 1: Differential pulse code modulation transmitter

From Fig (1), 𝑥(𝑛𝑇s ) represents the sampled version of the analog signal 𝑥(𝑡).
• The output of the comparator is the difference between the un-quantized sampled input
& prediction of its 𝑥ˆ(𝑛𝑇𝑠 ) i.e.
𝒆(𝒏𝑻𝒔 ) = 𝒙(𝒏𝑻𝒔 ) − 𝒙ˆ(𝒏𝑻𝒔 ) → (𝟏)
Where 𝑥ˆ(𝑛𝑇𝑠 ) is the prediction of 𝑥(𝑛𝑇𝑠 ).

• The prediction error 𝑒(𝑛𝑇𝑠 ) is quantized to produce 𝑒𝑞 (𝑛𝑇𝑠 ). In the quantizer, the noise
𝑞𝑒 (𝑛𝑇𝑠 ) gets added.

∴ The output of the quantizer can be written as:

𝒆𝒒 (𝒏𝑻𝒔 ) = 𝒆(𝒏𝑻𝒔 ) + 𝒒𝒆 (𝒏𝑻𝒔 ) → (𝟐)
• From Fig. (1), the input to the prediction fitter maybe written as:

𝒙𝒒 (𝒏𝑻𝒔 ) = 𝒙ˆ(𝒏𝑻𝒔 ) + 𝒆𝒒 (𝒏𝑻𝒔 ) → (3)

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Substituting Eq (2) in Eq (3), we get

𝒙𝒒 (𝒏𝑻𝒔 ) = 𝒙ˆ(𝒏𝑻𝒔 ) + 𝒆(𝒏𝑻𝒔 ) + 𝒒𝒆 (𝒏𝑻𝒔 ) → (4)

Substituting Eq (1) in Eq (4), we get

𝑥𝑞 (𝑛𝑇𝑠 ) = 𝑥ˆ(𝑛𝑇𝑠 ) + 𝒙(𝒏𝑻𝒔 ) − 𝒙ˆ(𝒏𝑻𝒔 ) + 𝑞𝑒 (𝑛𝑇𝑠 )
𝒙𝒒 (𝒏𝑻𝒔 ) = 𝒙(𝒏𝑻𝒔 ) + 𝒒𝒆 (𝒏𝑻𝒔 )

Where 𝑥𝑞 (𝑛𝑇𝑠 ) is the quantized version of 𝑥(𝑛𝑇𝑠 ).

DPCM Receiver:

Fig. 2: DPCM receiver

• The decoder 1st reconstruct the quantized error signal from incoming binary signal.
• The prediction filter output & quantized error signals are summed up to give the
quantized version of the original signal.
• Thus, the Signal at the receiver differs from the actual Signal by quantization error
𝑞𝑒 (𝑛𝑇𝑠 ), which is introduced permanently in the reconstructed Signal.

Prediction gain:
• The output signal to noise Ratio of a DPCM System is
𝝈𝟐𝒙
( SNR )𝟎 = 𝟐
𝝈𝑸𝟐
Where 𝜎𝑥 is the variance of the original input 𝑥(𝑛𝑇𝑠 ), assumed to be of zero means, & 𝜎𝑄2 is the
2

variance of the quantization error 𝑞(𝑛𝑇𝑠 ).

Eq (1) can be rewritten as
𝜎𝑥2 𝜎𝐸2
(𝑆𝑁𝑅)0 = 2 ⋅ 2
𝜎𝑄 𝜎𝐸
𝜎𝑥2 𝜎𝐸2
= 2⋅ 2
𝜎𝐸 𝜎𝑄
(𝑆𝑁𝑅)0 = 𝐺𝑃 ⋅ (S𝑁𝑅)𝑃
Where (𝑆𝑁𝑅)𝑝 is the prediction error to quantization noise ratio defined by
𝜎𝐸2
(𝑆𝑁𝑅)𝑃 = 2 &
𝜎𝑄
𝐺𝑝 is the prediction gain defined by
𝝈𝟐𝒙
𝑮𝒑 = 𝟐
𝝈𝑬

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

DELTA MODULATION (DM):

Delta modulation transmits only one bit per Sample i.e. the present sample value is compared
with the previous sample value & the indication, whether the amplitude is Increased or decreased
is sent.

Fig. 1: Illustration of Delta Modulation

• The input Signal 𝑥(t) is approximated to step signal by the delta modulater. The difference
between input signal 𝑥(𝑡) & staircase approximated signal is quantized into only two
levels i.e. +δ and -δ.
• If the difference is +ve, then approximated signal is increased by one step i.e. +𝜕 & bit 1 s
transmitted.
• If the difference is -ve, then approximated signal is reduced by one step, i.e. − ∂ & bit 0 is
transmitted.
• Thus far each sample, only one bit is transmitted.

Fig. 2: Step Size delta

DM Transmitter:

Fig. 3: DM Transmitter

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

• The error between the Sampled value 𝑥(𝑛𝑇𝑠 ) & last approximated Sample is given by

𝒆(𝒏𝑻𝒔 ) = 𝒙(𝒏𝑻𝒔 ) − 𝒙ˆ(𝒏𝑻𝒔 ) → (𝟏)

• Let 𝑢(𝑛𝑇𝑠 ) be the present sample approximation of staircase output.

From Fig. 3
𝒙ˆ(𝒏𝑻𝒔 ) = 𝒖(𝒏 − 𝟏)𝑻𝒔
𝒙ˆ(𝒏𝑻𝑺 ) = 𝒖(𝒏𝑻𝑺 − 𝑻𝑺 ) → (𝟐)
Substituting equation (2) in equation (1), we get
𝒆(𝒏𝑻𝒔 ) = 𝒙(𝒏𝑻𝒔 ) − 𝒖(𝒏𝑻𝒔 − 𝑻𝒔 ) → (𝟑)
• The binary quantity 𝑏(𝑛𝑇𝑠 ) is the algebraic sign of the error (𝑛𝑇𝑠 ) , except for the scaling
factor 𝜕. i.e.
𝒃(𝒏𝑻𝒔 ) = 𝛛𝐬𝐠𝐧[𝒆(𝒏𝑻𝒔 )]
𝑏(𝑛𝑇𝑠 ) depends on the Sign of error 𝑒(𝑛𝑇𝑠 ), the Sign of Step-Size ' 𝜕 ' will be decided i.e.
𝒃(𝒏𝑻𝒔 ) = +𝛛, if 𝒙(𝒏𝑻𝒔 ) ⩾ 𝒙ˆ(𝒏𝑻𝒔 )
𝒃(𝒏𝑻𝒔 ) = −𝛛, if 𝒙(𝒏𝑻𝒔 ) ⩽ 𝒙ˆ(𝒏𝑻𝒔 )
• If 𝑏(𝑛𝑇𝑠 ) = + ∂, then binary ' 1 ' is transmitted
• If 𝑏(𝑛𝑇𝑠 ) = − ∂, then binary ' 0 ' is transmitted.
∴ 𝑢(𝑛𝑇𝑠 ) = 𝑢[𝑛𝑇𝑠 − 𝑇𝑠 ] + 𝑏(𝑛𝑇𝑠 )
• The pervious sample approximation 𝑢[𝑛𝑟𝑠 − 𝑇𝑠 ] is restored by delaying one sample period
' 𝑇𝑠 '.

DM Receiver:

Fig. 4: DM receiver
Fig. 4 shows the block diagram of DM Receiver.
• The accumulator generates the Staircase approximated signal output & is delayed by one
sampling period 𝑇𝑠 . It is then added to the input signal.
• If input is binary ' 1 ' then it adds to step +δ to the previous output.
• If input is binary '0' then one step ' ∂ ' is subtracted from the delayed signal.
• The LPF is used to remove step variation & to get smooth reconstructed message
signal 𝑥(𝑡).

Advantages of DM:
The DM has following advantages over PCM
1 𝐷𝑀 Transmits only one bit for one Sample. Thus, the signaling rate & transmission
channel bandwidth is relatively small for DM.
2 Simplicity of design for both the transmitter & the receiver.
3 A one-bit code word for the output, which eliminates the need for word framing.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Drawbacks of DM:
DM Systems are subjected to two types of quantization error:
1 Slope overload distortion
2 Granular Noise.

Fig. 1: Illustration of Quantization error in delta modulation

Slope overload distortion:
• Slope overload distortion arises because of the large dynamic range of the input signal.
• In Fig it can be Seen, the rate of rise of input signal 𝑥(𝑡) is so high that the staircase signal
can not approximate it, the step Size ' 𝜕 ' becomes too Small for staircase signal 𝑥(𝑡) to
follow the steep segment of 𝑥(𝑡). Thes large error between the staircases approximated
signal & the original input signal 𝑥(𝑡). This error is called Slope overload distortion.
• To reduce this error, the step size should be increased when the slope of the signal 𝑥(𝑡) is
high.
i.e., Slope of the Staircase 𝑢(𝑡) ⩾ Slope of the message Signal.

𝛛 𝒅
⩾ 𝐦𝐚𝐱 | [𝒙(𝒕)]|
𝑻𝒔 𝒅𝒕

Granular Noise:
• This noise occurs when the step Size is too large compared to small variations in the input
Signal i.e. for very small variations in the input Signal, the Staircase signal is changed by
large amount because of large Step Size ' 𝑑 '.
• In Fig , Input Signal is almost flat, the staircase Signal u(t) keeps on oscillating by ± ∂
around the signal.
• The error between the Input & approximated Signal is called Granular Noise. The Solution
of this Problem is to make Step size Small.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

PCM & DPCM FORMULAE

1 Number of levels : 𝐿 = 2𝑁
2 Number of bits : 𝑁 = log 2 (𝐿)
log10 (𝐿)
𝑁=
log10 (2)
3 Sampling rate : 𝑓𝑆 ⩾ 2𝑤
Where ' 𝑊 ' is the highest frequency of message
Signal
4 Nyquist rate : 𝑓𝑆 = 2𝑊
5 Signaling rate or Bit : 𝑟𝑏 = 𝑁𝑓𝑠 or 𝑟𝑏 = 𝑁 ⋅ 2𝑊
transmission rate
6 Transmission : 1 1
𝐵𝑇 = 𝑁𝑓𝑆 𝐵𝑇 = 𝑁 ⋅ 2𝑊
bandwidth 𝐵𝑇 2 2
1
𝐵𝑇 = 𝑟𝑏 𝐵𝑇 = 𝑁𝑊
2
Where 𝑟𝑏 = 𝑁𝑓𝑆
7 Number of bits : 𝑟𝑏
𝑁=
𝑓𝑠
8 Bit rate : 𝑅𝑏 = 𝑁𝑓𝑠
9 Bit duration : 𝑇𝑏 = 1/𝑅𝑏
10 Sampling frequency : 𝑓𝑠 = 𝑅𝑏 /𝑁
11 Message bandwidth : 𝑊 = 𝑓5 /2
12 Maximum Signal to : 3P
quantization Ratio (SNR)0 = 2 ⋅ 22𝑁
𝑥max

Where 𝑥max = Maximum amplitude of message

Signal.
𝑃 = Signal power

𝑵𝒐𝒕𝒆: 𝒙𝐦𝐚𝐱 or 𝒗𝐦𝐚𝐱 or 𝒈𝐦𝐚𝐱

13 Signal power : 𝐴2𝑚
𝑃=
2
14 Quantization Noise :
∂2
power of Quantization 𝜎𝑄2 =
error of Normalized 12
Noise power
𝑁𝑜𝑡𝑒: ∂ or 𝛥
15 Step Size : 𝑥max − (−𝑥max )
∂=
𝐿
2𝑥max
∂=
2𝑁
16 Normalized Signal to : ( SNR )𝑑𝐵 = (4 ⋅ 8 + 6𝑁)
quantization noise ratio
17 Signal to quantization : (𝑆𝑁𝑅)𝑑𝐵 = 1.76 + 6 N
Noise ratio for
Sinusoidal signal

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

18 (𝑆𝑁𝑅) : 𝑃 𝐴2𝑚 /2
(𝑆𝑁𝑅) = =
𝜎𝑄2 Δ2 /12
19 Bit duration : 𝑇𝑆
𝑇𝑏 =
𝑁
𝑇𝑠 = 𝑁𝑇𝑏
20 (SNR)dB for Non- : (𝑆NR)dB = 4 ⋅ 8 + 6 N
Sinusoidal Signal
(example: Telephone
Signal)

Problem 11.
A Signal of BW 3.5 KHz is Sampled & Coded by a PCM System. The coded Signal is then transmitted
over a Channel at a rate of 50 Kbps. Calculate the maximum SNR that can be obtained by this
System. The input Signal has a peak value of 4𝑉.

Given: 𝑊 = 3.5kHz
Solution:

𝐴𝑚 = 𝑥max = 4V , 𝑟𝑏 = 50 Kbps, (𝑆𝑁𝑅)0 = ? , 𝑁= ?

Wkt
𝑓𝑠 = 2𝑊 = 2 × 3.5 KHz = 7 KHz
𝑟𝑏 = 𝑁𝐹𝑠
𝑟𝑏 50 Kbps
𝑁 = = = 𝟕. 𝟏𝟒𝟐
𝑓𝑠 7 KHz
Choose 𝑁 = 8
3𝑃22𝑁
(𝑆𝑁𝑅)0 = 2
𝑥max
𝐴2m 42
𝑃= = =𝟖𝑾
2 2
3 × 8 × 22X8
( SNR )0 = = 𝟗𝟖𝟑𝟎𝟒
(4)2
( SNR )𝑑8 = 10 log10 (98304)
( SNR )𝒅𝟖 = 𝟒𝟗. 𝟗 𝐝𝐁
II Method:
( SNR )𝒅𝟖 = 𝟏. 𝟖 + 𝟔𝐍 = 𝟏. 𝟖 + 𝟔 𝐗 𝟖 = 𝟒𝟗. 𝟖 𝐝𝐁

Problem 12.
A 10 KHz Sinusoid With amplitude 1 V peak is quantized to have SNR of about 45 dB. Find the
number of bits required per Sample, bit rate & bandwidth of the System if Sampling frequency is
twice the Nyquist rate.
Given: 𝑊 = 10 KHz, 𝐴𝑚 = 1 V, (SNR)dB = 45dB
Solution:
𝑁 =? , 𝑟𝑏 =? , 𝐵𝑇 = ?
Wkt
(𝑆𝑁𝑅)0 = 1.8 + 6 N
45 = 1.8 + 6 N
6 N = 45 − 1.8 = 43.2
43.2
𝑁 = = 𝟕. 𝟐
6
Choose 𝑵 = 𝟖

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

• Nyquist rate = 2W = 2 × 10 KHz = 20 KHz.

• Sampling frequency 𝑓s = 2𝑋 Nyquist rate
= 2 × 20 KHz
𝒇𝒔 = 𝟒𝟎 𝐊𝐇𝐳
• Bit rate
𝑟𝑏 = 𝑁𝑓𝑠
= 8 × 40 KHz
𝑟𝑏 = 𝟑𝟐𝟎 𝐊𝐛𝐩𝐬
• Bandwidth
1 320 Kbps
𝐵𝑇 = 𝑟𝑏 = = 𝟏𝟔𝟎 𝐊𝐇𝐳
2 2

OR
1 1
𝐵𝑇 = 𝑁𝑓𝑆 = 8 × 40 KHz = 160 KHZ
2 2

Problem 13.
A PCM System uses a uniform quantizer followed by a 7-bit encoder. The bit rate of the System is
equal to 50 × 106 bits/sec.
i. What is the maximum me sage bandwidth for which the System operates Satisfactorily?
ii. Determine the output Signal to quantization noise ratio when a full load Sinusoidal
modulating ware of frequency 1 MHz is applied to the input.
Given: 𝑁 = 7-bit, 𝑟𝑏 = 50 × 106 bits /sec. 𝑊 = ? , ( SNR )𝑑8 = ?
Solution:

i)
𝑟𝑏 = 𝑁𝑓𝑆
𝑟𝑏 = 𝑁 2𝑊
𝑟𝑏 50 × 106
𝑊= =
2𝑁 2×7
𝑊 = 3.57 MHz

ii)
(𝑆𝑁𝑅)𝑑𝐵 = 1.8 + 6 N
= 1.8 + 6 × 7
( SNR )𝒅𝑩 = 𝟒𝟑. 𝟖 𝐝𝐁

Problem 14.
A telephone Signal with bandwidth 4 KHz is digitized into an 8-bit PCM, Sampled at Nyquist rate.
Calculate PCM transmission bandwidth & Signal to quantization noise ratio (SNR).
Given: 𝑊 = 4 KHz, 𝑁 = 8 bit
𝐵𝑇 = ? & (𝑆𝑁𝑅)𝑑𝐵 = ?
Solution:
• PCM transmission bandwidth
1
𝐵𝑇 = 𝑁𝑓𝑆
2
𝐵𝑇 = 2 𝑊 = 2 × 4 KHz = 8 KHz
1
𝐵𝑇 = 8 × 8 KHz
2
𝑩𝑻 = 𝟑𝟐 𝐊𝐇𝐙

(OR)

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

𝐵𝑇 = 𝑁𝑊
𝐵𝑇 = 8 × 4 KHz
𝑩𝑻 = 𝟑𝟐 𝐊𝐇𝐳
• Telephone Signal is Non-Sinusoidal Signal
∴ (𝑆𝑁𝑅)𝑑𝐵 = 4.8 + 6 N
= 4.8 + 6 × 8
(𝑺𝑵𝑹)𝒅𝑩 = 𝟓𝟐. 𝟖 𝐝𝐁
Problem 15.
A PCM system which employs uniform quantization & produces a binary output given an input
Signal whose amplitude varies from +4 Volt to -4 volt & having average power of 40 mW. Calculate
the number of bits/Sample if the required Signal to Noise Ratio is 20 dB
Given: 𝑥max = 4 V, 𝑃 = 40 mW, (SNR) )dB = 20 dB, 𝑁 = ?
Solution:
• Wkt
(𝑆𝑁𝑅)𝑑𝐵 = 10 log10 (𝑆𝑁𝑅)
˙
20 𝑑𝐵 = 10 log10 (𝑆𝑁𝑅)
20
log10 (𝑆𝑁𝑅) =
10
−1
(𝑆𝑁𝑅) = log10 (2)

(𝑺𝑵𝑹) = 𝟏𝟎𝟎
• Wkt
3 𝑃 22𝑁
(𝑆𝑁𝑅) = 2
𝑥max
3 × 40 × 10−3 × 22𝑁
100 =
(4)2
100 × 16
22𝑁 = WKT
3 × 40 × 10−3 2𝑁 = 𝐿
22𝑁 = 13333.33 𝑁 = log 2 (𝐿)
2𝑁 = log 2 (13333.33) Where
𝐥𝐨𝐠 𝟏𝟎 (𝟏𝟑𝟑𝟑𝟑. 𝟑𝟑) 𝐥𝐨𝐠 𝟏𝟎 (𝑳)
𝟐𝑵 = 𝐥𝐨𝐠 𝟐 (𝑳) =
𝐥𝐨𝐠 𝟏𝟎 (𝟐) 𝐥𝐨𝐠 𝟏𝟎 (𝟐)
2𝑁 = 13.70274
13.70274
𝑁 =
2
𝑵 = 𝟔. 𝟖𝟓

Choose

𝑵=𝟕

(𝑆𝑁𝑅)𝑑𝐵 = 10log10 (36922.84)

(𝑺𝑵𝑹)𝒅𝑩 = 𝟒𝟓. 𝟔𝟕 𝐝𝐁

Problem 16.
A 10 KHz Sinusoid with amplitude levels of ±1 Volt is to be Sampled & quantized by rounding off.
How many bits are required to ensure a quantization SNR of 45 dB ? What is the bit rate of the
digitized Signal if the Sampling rate is chosen as twice the Nyquist rate?

Given: 𝐴𝑚 = ±1 V, (S𝑁𝑅)𝑑𝐵 = 45 dB, 𝑊 = 10KHZ.

Solution:

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

(𝑆𝑁𝑅)𝑑𝐵 = 1.76 + 6 N
45 = 1.76 + 6 N
6N = 43.24
𝑵 = 𝟕. 𝟐𝟎𝟔
Choose
𝑵=𝟖
• Nyquist rate = 2W = 2 × 10 KHz = 𝟐𝟎 𝐊𝐇𝐳
• Sampling rate 𝑓𝑠 = 2 × Nyquist rate = 2 × 20 KHZ
𝒇𝐬 = 𝟒𝟎 𝐊𝐇𝐳
• Bit rate 𝑅𝑏 = 𝑁𝑓𝑠 = 8 × 40 KkHz
𝑹𝒃 = 𝟑𝟐𝟎 𝐊𝐛𝐩𝐬

TIME DIVISION MULTIPLEXING (TDM):

• An Important feature of TDM is Conservation of time i.e. different time Intervals (periods)
are allocated for different message Signals, So that a Common Channel is utilized for
transmission of these Signals without any Interference.

Fig (1): Block diagram of TDM system.

The concept of TDM is illustrated in the block diagram.
• The Low pass pre-alias filters are used to remove high-frequency components which may
be present in the message signal.
• The output of the pre-alias filters are then fed to a commutator, which is usually
implemented using electronic Switching circuitry.
• The function of Commutator is 2-fold:
1. To take a Narrow Sample of each of the " 𝑁 ' Input Signals at a rate 𝑓𝑆 ⩾ 2𝑊, where
' 𝑤 ' is the cut-off frequency of pre-alias filter.
2. To Sequentially interleave these ' 𝑁 ' Samples inside a Sampling interval 𝑇𝑠 = 1/𝑓𝑠 .
This Interleaving is nothing but multiplexing.
• The multiplexed Signal is applied to a pulse amplitude modulator whose purpose is to
transform the multiplexed Signal into a form Suitable for transmission over a common
Channel.
• At the receiving end, the pulse amplitude demodulator performs the reverse operation of
PAM & the de-commutator distributes the signals to the appropriate low pass re-
construction filters.
• The de-commutator operates in synchronization with the commutator.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Fig. 2: Waveforms illustrating TDM for two message signals.

• Suppose that the ' 𝑁 ' message Signals to be multiplexed (Txed) have the same Spectral
properties (BW). Then the sampling rate for each message Signal is determined in
accordance with the Sampling theorem.
• Let 'Ts' denote the Sampling period.
Let ' 𝑇𝑥 ' denote the time Spacing between adjacent samples in the TDM Signal.
𝑇
i.e. 𝑇𝑥 = 𝑁𝑠 as Shown in fig (2).

NOTE:
𝑇
• Spacing between two Samples ' 𝑇𝑥 ' = 𝑆
𝑁
1 1 𝑁
• Number of pulses per Second = 𝑇 = 𝑇 = 𝑇 = 𝑁𝑓𝑆
𝑥 𝑆 /𝑁 𝑆
• Number of pulse per second is also called as Signaling rate' 𝑟 ′ i.e. r = 𝑁𝑓𝑆
Since 𝑓𝑆 ⩾ 2𝑓𝑚
r ⩾ 𝑁2𝑓𝑚
Signalling Rate
• Transmission Bandwidth =
2

TDM FORMULAE

1. Speed of the Commutator in revolution per Second (rps) = 2𝑊 Where ′𝑊′ is the minimum
BW of message signal.
2. Speed of the Commutates in Samples/ Sec
= Total number of segments 𝑋 speed of Commutator in 'rps'
1
3. Minimum transmission 𝐵𝑊 = 2 [ Sum of Nyquist Rate ]
1
4. Minimum bit rate = 2 [ Sum of Nyquist rate ] bits /sec.
360∘
5. Angle of Separation of Corresponding segments = 𝑁
Where ' 𝑛 ' is the number of Segments.
360∘
6. Angle of Separation b/w each Segment ( pole ) =
Total Number of Segments

Problem 17:
Signal 𝑚1 (𝑡) is band-limited to 3𝐾𝐻𝑧 & 3 other Signals 𝑚2 (𝑡), 𝑚3 (𝑡) & 𝑚4 (𝑡) are bandimited to
1.5 kHz each. These are transmitted by means of TDM.
i. Set up a commutator scheme to realize the multiplexing with each Signal Sampled at
Nyquist rate.
ii. Find the speed of the commutator in Samples/sec & the minimum band-with of the channel

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Solution
i)
Nyquist Angle of Separation of Corresponding
Message No of
BW rate 360∘
Signal Segments'W' Segments =
𝑓s = 2 W 𝑁
𝑚1 (𝑡) 3KHz 6KHz 2 180∘
𝑚2 (𝑡) 1.5KHz 3KHz 1 360∘
𝑚3 (𝑡) 1.5KHz 3KHz 1 360∘
𝑚1 (𝑡) 1.5KHz 3KHz 1 360∘

360∘
• Angle of separation b/w each segment ( pole ) =
Total No of Segments
360∘
= = 𝟕𝟐∘
5

• If the commutator is rotated at 3000 rev/sec, then in each revolution we obtain one
Sample each for 𝑚2 (𝑡), 𝑚3 (𝑡)&𝑚4 (𝑡)& 2 Samples from 𝑚1 (𝑡).
• commutator speed in rps = 2𝑊 = 2 × 1.5kHz = 3000rps
ii) Speed of the commutator in samples/sec
= [ Total No of Segments 𝑥 speed of Commutator in rps ]
= 5 × 3000 rps
= 𝟏𝟓, 𝟎𝟎𝟎 Samples/sec.
1
Minimum BW of Channel = 2 [ Sum of Nyquist rate ]
= 1/2[6KHz + 3KHz + 3KHz + 3KHz]
= 15 kHz/2
= 𝟕. 𝟓 𝐤𝐇𝐳.

Problem 18:
A Signal 𝑚1 (𝑡) is band limited to 3.6 KHz & three other Signals 𝑚2 (𝑡), 𝑚3 (𝑡) & 𝑚4 (𝑡) core band
limited to 1.2 KHz each. There Signals are to be transmitted by means of TDM. Sketch set up a
Scheme for realizing this multiplexing requirement with each Signal Sampled at its Nyquist rate.
Determine the Speed of Commutator in Samples per Second.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Solution:

Nyquist No. of Angle of Separation of 360∘

Message
BW rate Segments corresponding =
Signal 𝑁
𝑓s = 2 W 'N' Segments
𝑚1 (𝑡) 3.6KHz 7.2KHz 3 120∘
𝑚2 (𝑡) 1.2KHz 2.4KHz 1 360∘
𝑚3 (𝑡) 1.2KHz 2.1KHz 1 360∘
𝑚4 (𝑡) 1.2KHz 2.4KHz 1 360∘

360∘
• Angle of separation b/w each segment ( pole ) = Total No of Segments
360∘
= = 𝟔𝟎∘
6

• If the Commutator is rotated at 2400 rev/sec then in each revolution, we obtain one
sample each for 𝑚2 (𝑡), 𝑚3 (𝑡) & 𝑚4 (𝑡) and 3 Samples from 𝑚1 (𝑡).
• Commutator speed in rps = 2𝑊 = 2 × 1.2KHz = 𝟐𝟒𝟎𝟎𝐫𝐩𝐬
• Speed of Commutator in Samples/sec
= Total No of segments × Speed of commutation in rps.
= 6 × 2400
= 𝟏𝟒, 𝟒𝟎𝟎𝐒𝐚𝐦𝐩𝐥𝐞𝐬/𝐒𝐞𝐜.
1
Minimum BW of the Channel = 2 [7.2KHz + (3 × 2.4kHz)] = 𝟕. 𝟐 𝐊𝐇𝐳.

Digital Multiplexers:

Fig. 1: Multiplexing and Demultiplexing

• Multiplexing of digital Signals is also possible when the data Sources are operating at
different bit rate.
• Multiplexing is done to Combine Several digital Signal such as Computer output, digitized
telephone Signals, digitized TV Signals etc. into a Single data Streams.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

• The multiplexing of digital Signal may be accomplished by using a bit by bit interleaving
procedure with a Selector switch that Sequentially takes a bit from each incoming line &
then applies to the high-speed common line.
• At the receiving end of the System the o/p of this common line is Separated out into its
low-Speed individual Components & then deliver to their respective destinations.

Two major groups of digital multiplexers core used in practice:

• One group of multiplexers is designed to combine relatively low Speed digital Signals, up
to a maximum rate of 4800 bps into a higher Speed multiplexed Signal with a rate of up to
9600 bps
• Used to transmit data over voice-grade Channels of a telephone Network.

T1 carrier system or Digital Multiplex T1 or TDM PCM Telephony or Digital

Hierarchy (T1 to T4 carrier system):

Fig. 1: T1 carrier system

Fig. 2: T1 carrier frame

• The 𝑇1 carrier system is designed to accommodate 24 voice Channels, primarily for Short-
distance, heavy usage in metropolitan areas.

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

• This System was developed by the Bell System in the United States in the early 1960's. The
𝑇1 System has been adopted for use throughout the United States, Canada & Japan.
• voice signal is bandlimited to 300 Hz to 3400 Hz even through the Nyquist rate is 6.8KHz,
the Standard Sampling rate used in digital telephony is 8000 Sampler/sec.
• So 24 voice Signals are Sampled at a rate of 8KHz & the Resulting Samples are quantized
& converted into 7-bit PCM codewords.
• At the end of each 7-bit codeword, an additional binary bit is added for synchronizing
purpose. At the end of every group of twenty-four 8-bit Codewords, another additional bit
is inserted to give frame Synchronization.
• The overall frame size in the 𝑇1 -Carrier is
𝑇1 = [(24 voice Signals × 8-bit ) + 1-bit ] bits/frame
𝑇1 = (192 + 1) bits/frame
𝑻𝟏 = 𝟏𝟗𝟑 bits/frame
• These frames are transmitted at 8KHz rate
∴ Bit rate = 193 bits / frame × 8000 frames /sec
Bit rate = 𝟏. 𝟓𝟒𝟒 𝐌𝐛𝐩𝐬
• Sampling period 𝑻𝒔 = 𝟏/𝒇𝒔 = 𝟏/𝟖𝟎𝟎𝟎 = 𝟏𝟐𝟓µ𝐬𝐞𝐜
• For 193 bits, time period is 125µsec
𝟏𝟐𝟓µ𝐬𝐞𝐜
∴ Duration of each bit = 𝟏𝟗𝟑
= 𝟎. 𝟔𝟒𝟕µ𝐬𝐞𝐜
• The maximum length of the 𝑇1 system is now limited to 50 to 100 miles with a repeater
Spacing of 1-Mile.
• The overall T1 carrier system is designed for accommodating voice Channels, picture
phone service, TV Signals & digital data.

DELTA MODULATION FORMULAE

1 Slope overload distortion will occur if

∂
𝐴𝑚 ≥
2𝜋𝑓𝑚 𝑇𝑠
2 Slope overload distortion will not occur if
∂
𝐴𝑚 ≤
2𝜋𝑓𝑚 𝑇𝑠
3 Nyquist rate = 2 W
Where ' 𝑊 ' → Maximum frequency of the Signal.
4 Sampling Interval 𝑇𝑆 = 1/𝑓𝑆
5 𝑹𝒃 in bits/𝑺𝒆𝒄 = 𝒇𝒔 in Samples /𝑺𝒆𝒄.
example:
Bit rate = 20 Kbit/sec Bit rate = 60 K bits/Sec
Sampling rate 𝑓𝑠 = 20 Kilo Samples/Sec Sampling rate 𝑓𝑠 = 60 Kilo Samples /sec

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Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Problem 19.
Consider a Speech Signal with a maximum frequency of 3.4KHz & maximum amplitude of 1V. The
Speech Signal is applied to a DM with its bit rate at 20kbit/sec.

Given: 𝑊 = 𝑓0 = 3.4kHz, 𝐴𝑚 = 1 V, 𝑟𝑏 = 20 Kbps = 𝑓𝑆 = 20 K Samples /sec.

Solution:
Wkt the Slope overload distortion will not occur if
∂
𝐴𝑚 ≤
2𝜋𝑓0 𝑇𝑠
∂ ≥ 𝐴𝑚 2𝜋𝑓0 𝑇𝑠
1
∂ ≥ 𝐴𝑚 2𝜋𝑓0
𝑓𝑠
1
∂ ≥ 1 × 2𝜋 × 3.4 × 103
20 K Samples/sec
𝛛 ≥ 𝟏. 𝟎𝟔𝟖 𝐕
Problem 20.
Consider a Speech Signal with maximum frequency of 3.4 KHz & maximum amplitude of 1 Volt.
This speech Signal is applied to a delta modulator whose bit rate is Set at 60 Kbit/sec.
Given: 𝑊 = 𝑓0 = 3.4 KHz, 𝐴𝑚 = 1 V, 𝑅𝑏 = 60 Kbps = 𝑓𝑆 = 60 K Samples/sec.
Solution:
Wkt the Slope overload distortion will not occur if

∂
𝐴𝑚 ≤
2𝜋𝑓0 𝑇𝑠
∂ ≥ 𝐴𝑚 2𝜋𝑓0 𝑇𝑠
1
∂ ≥ 1 × 2𝜋 × 3.4 × 103 ×
60 × 103
𝛛 ≥ 𝟎. 𝟑𝟓𝟔 volts

Problem 21.
Assume a speech Signal with a minimum frequency of 3.4kHz & a maximum amplitude of 1V. The
speech signal is applied to a delta modulates with its bit hate at 25kaps. Diffuse the Choice of an
appropriate Step Size for the delta modulates.

Given: 𝑓0 = 3.4kHz, 𝐴𝑚 = 1 V, 𝑅𝑏 = 25kkps = 𝑓𝑆 = 25 K Samples/sec

Solution:

Wkt Slope overload distortion will not occur if

∂
𝐴𝑚 ≤
2𝜋𝑓0 𝑇𝑠
1
∂ ≥ 𝐴𝑚 2𝜋𝑓0 𝑇𝑠 ≥ 𝐴𝑚 2𝜋𝑓0
𝑓𝑆
1
∂ ≥ 1 × 2𝜋 × 3.4 × 103 ×
25 × 103
𝛛 ≥ 𝟎. 𝟖𝟓𝟓 Volts

Page | 41
Communication Engineering Unit-4: Sampling, PCM, DPCM, DM & TDM

Comparison of digital pulse modulation methods:

Differential Pulse
Sl. Delta modulation
Parameter PCM Code Modulation
No. (DM)
(DPCM)

It can use 4,8 Bits can be It uses only

Number of or 16 bits per more than one
1. one bit for one
Bits but are less
sample. sample.
than PCM.

The number
of levels Step size is
Levels, step depend on Fixed number of fixed and
2
size number of levels are used. cannot be
bits. Level varied.
size is fixed.

Slope
Quantization Slope overload
Quantization overload
error depends distortion and
3 error and distortion and
on number of quantization
Distortion granular noise
levels used noise is present.
is present.

Highest
Bandwidth bandwidth is Lowest
Bandwidth
of trans- required since bandwidth is
4 required is lower
mission required.
number of than PCM.
channel
bits are high.

There is no
Feedback
feedback in
5 Feedback Feedback exists. exists in
transmitter or
transmitter.
receiver.

System System is
6 Simple Simple
complexity complex

7 SNR Good Fair Poor

Audio &
Speech & Speech &
8 Applications video
video Images
telephony

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