Program : B.

Tech
Subject Name: Digital Communication
Subject Code: EC-502
Semester: 5th
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Department of Electronics and Communication Engineering

Sub. Code: EC 5002 Sub. Name: Digital Communication

Unit 2
Digital conversion of Analog Signals: Sampling theorem, sampling of band pass signals, Pulse Amplitude
Modulation (PAM), types of sampling (natural, flat-top), equalization, signal reconstruction and
reconstruction filters, aliasing and anti-aliasing filter, Pulse Width Modulation (PWM), Pulse Position
Modulation (PPM)
Digital transmission of Analog Signals: Quantization, quantization error, Pulse Code Modulation (PCM),
companding, scrambling, TDM-PCM, Differential PCM, Delta modulation, Adaptive Delta modulation,
vocoder.

PART I DIGITAL CONVERSION OF ANALOG SIGNALS

2.1 Sampling of Analog Signals

Sampling is defined as, “The process of measuring the instantaneous values of continuous-time signal in a
discrete form.” In the process of sampling an analog signal is converted into a corresponding sequences of
samples, that are uniformly spaced in time.
This discretization of analog signal is called as Sampling. The following figure indicates a continuous-time
signal x (t) and a sampled signal xs (t). When x (t) is multiplied by a periodic impulse train, the sampled
signal xs (t) is obtained.

Figure 2.01 Sampling

2.1.1 Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be termed as a
sampling period 𝑇𝑠.
𝑆𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝐹𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦 = 1/𝑇𝑠 = 𝑓𝑠
Where,
𝑇𝑠 = 𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑡𝑖𝑚𝑒
𝑓𝑠 = 𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑓𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦 𝑜𝑟 𝑡ℎ𝑒 𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑟𝑎𝑡𝑒

Sampling frequency is the reciprocal of the sampling period. This sampling frequency can be simply called
as Sampling rate. The sampling rate denotes the number of samples taken per second, or for a finite set of
values.

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For an analog signal to be reconstructed from the digitized signal, the sampling rate should be highly
considered. The rate of sampling should be such that the data in the message signal should neither be lost
nor it should get over-lapped. Hence, a rate was fixed for this, called as Nyquist rate.

2.1.2 Signals Sampling Techniques

There are three types of sampling techniques:
a. Impulse sampling.
b. Natural sampling.
c. Flat Top sampling.

(a) Impulse Sampling

Impulse sampling can be performed by multiplying input signal x(t) with impulse train of period 'T'. Here,
the amplitude of impulse changes with respect to amplitude of input signal x(t). The output of sampler is
given by

Figure 2.1.2.1 Impulse Sampling

𝑦(𝑡) = 𝑥(𝑡) × 𝑖𝑚𝑝𝑢𝑙𝑠𝑒 𝑡𝑟𝑎𝑖𝑛

∞

= 𝑥(𝑡) × ∑ 𝑥(𝑛𝑡)𝛿 (𝑡 − 𝑛𝑡)

𝑛=−∞
∞

𝑦(𝑡) = 𝑦𝑛 (𝑡) = ∑ 𝑥(𝑛𝑡)𝛿 (𝑡 − 𝑛𝑡) … …(2.1.2.1)

𝑛=−∞
To get the spectrum of sampled signal, consider Fourier transform of equation 1 on both sides
∞
1
𝑌(𝜔) = ∑ 𝑋(𝜔 − 𝑛𝜔𝑠 )
𝑇0
𝑛=−∞
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.
(b) Natural Sampling
Natural sampling is similar to impulse sampling, except the impulse train is replaced by pulse train of
period T. i.e. you multiply input signal x(t) to pulse train ∑∞
𝑛=−∞ 𝑃(𝑡 − 𝑛𝑇 ) as shown below

Figure 2.1.2.2 Natural Sampling

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𝑦(𝑡) = 𝑥(𝑡) × 𝑝𝑢𝑙𝑠𝑒 𝑡𝑟𝑎𝑖𝑛

= 𝑥(𝑡) × 𝑝(𝑡)
∞

= 𝑥(𝑡) × ∑ 𝑃(𝑡 − 𝑛𝑇) …(2.1.2.2)

𝑛=−∞

The exponential Fourier series representation of p(t) can be given as

∞

𝑝(𝑡) = ∑ 𝐹𝑛 𝑒 𝑗𝑛𝜔0 𝑡 …(2.1.2.3)

𝑛=−∞

= ∑ 𝐹𝑛 𝑒 𝑗2𝜋𝑛𝑓0 𝑡
𝑛=−∞

𝑇
1
Where 𝐹𝑛 = 𝑇 ∫2𝑇 𝑝(𝑡)𝑒 𝑗𝑛𝜔0 𝑡 𝑑𝑡
−
2

1
= (𝑛𝜔𝑠 )
𝑇𝑃
Substitute Fn value in equation 2.1.2.2
∞
1
∴ p(t) = ∑ P(nωs )ejnω0 t
T
n=−∞
∞
1
= ∑ P(nωs )ejnω0 t
T
n=−∞

Substitute p(t) in equation 2.1.2.1

𝑦(𝑡) = 𝑥(𝑡) × 𝑝(𝑡)
∞
1
= 𝑥(𝑡) × ∑ 𝑃(𝑛𝜔𝑠 )𝑒 𝑗𝑛𝜔0 𝑡
𝑇
𝑛=−∞

∞
1
= ∑ 𝑃(𝑛𝜔𝑠 )𝑥(𝑡)𝑒 𝑗𝑛𝜔0 𝑡
𝑇
𝑛=−∞

To get the spectrum of sampled signal, consider the Fourier transform on both sides.
∞
1
𝐹. 𝑇. [y(t)] = F. T. ∑ [P(nωs )x(t)ejnω0 t ]
T
n=−∞
∞
1
= ∑ P(nωs )F. T. [x(t)ejnω0 t ]
T
n=−∞

According to frequency shifting property

F. T. [x(t)ejnω0 t ] = X[𝜔 − 𝑛𝜔𝑠 ]
∞
1
∴ Y[𝜔] = ∑ P(nωs ) X[𝜔 − 𝑛𝜔𝑠 ]
T
n=−∞

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(c) 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.

Figure 2.1.2.3 Flat Top Sampling

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:
i.e. 𝑦(𝑡) = 𝑝(𝑡) × 𝑦𝛿 (𝑡) …(2.1.2.4)

Figure 2.11 Sampling

To get the sampled spectrum, consider Fourier transform on both sides for equation 1
𝑌[𝜔] = 𝐹. 𝑇. [𝑃(𝑡) × 𝑦𝛿 (𝑡)]
By the knowledge of convolution property,
𝑌[𝜔] = 𝑃 (𝜔)𝑌𝛿 (𝜔)
𝜔𝑇
Here 𝑃(𝜔) = 𝑇𝑆𝑎 ( ) = 2𝑠𝑖𝑛𝜔𝑇/𝜔
2

2.1.3 Sampling Theorem

The sampling theorem states that, “a signal whose spectrum is band limited to B Hz, [G(ω) = 0 for |𝜔| >
2𝜋𝐵] can be reproduced exactly from its samples if it is sampled at the rate 𝑓𝑠 which is greater than twice
the maximum frequency ω of the signal to be sampled.” Therefore minimum sampling frequency is
𝑓𝑠 = 2𝐵 𝐻𝑧
Proof: Consider a continuous time signal x(t). The spectrum of x(t) is a band limited to f m Hz i.e. the
spectrum of x(t) is zero for |ω|>ωm.
Sampling of input signal x(t) can be obtained by multiplying x(t) with an impulse train δ(t) of period Ts. The
output of multiplier is a discrete signal called sampled signal which is represented with y(t) in the following
diagrams:

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Figure 2.1.3.1 Sampling of signal x(t)

Here, you can observe that the sampled signal takes the period of impulse. The process of sampling can be
understood as under.
The sampled signal is given by
𝑦(𝑡) = 𝑥(𝑡). 𝛿 (𝑡) = 𝑥(𝑡). 𝛿𝑇𝑠 (𝑡)
= ∑ 𝑥(𝑛𝑇𝑠 ). 𝛿(𝑡−𝑛𝑇𝑠 ) ….(2.1.3.1)
𝑛
The impulse train 𝛿𝑇𝑠 (𝑡) is a periodic signal of period 𝑇𝑠 , hence it can be expressed as a Fourier series
1
𝛿𝑇𝑠 (𝑡) = [1 + 2 𝑐𝑜𝑠𝜔𝑠 𝑡 + 2 𝑐𝑜𝑠2𝜔𝑠 𝑡 + 2 𝑐𝑜𝑠3𝜔𝑠 𝑡 + ⋯ ]
𝑇𝑠
2𝜋
Where 𝜔𝑠 = 𝑇 = 2𝜋𝑓𝑠 ….(2.1.3.2)
𝑠

Substitute 𝛿 (𝑡)in equation 1.

→ 𝑦 (𝑡 ) = 𝑥 (𝑡 ). 𝛿 (𝑡 )
1
𝑦(𝑡) = [𝑥(𝑡) + 2𝑥(𝑡) 𝑐𝑜𝑠𝜔𝑠 𝑡 + 2𝑥(𝑡) 𝑐𝑜𝑠2𝜔𝑠 𝑡 + 2 𝑥(𝑡)𝑐𝑜𝑠3𝜔𝑠 𝑡 + ⋯ ]
𝑇𝑠
Now to find the 𝑌(𝜔), we have to take the fourier transform of both the sides,
1
𝑌(𝜔) = [𝑋(𝜔) + 𝑋(𝜔 − 𝜔𝑠 ) + 𝑋(𝜔 + 𝜔𝑠 ) + 𝑋(𝜔 − 2𝜔𝑠 ) + 𝑋(𝜔 + 2𝜔𝑠 ) + ⋯ ]
𝑇𝑠
1
𝑌(𝜔) = 𝑇 ∑∞ 𝑛=−∞ 𝑋(𝜔 − 𝑛𝜔𝑠 ) Where 𝑛 = 0, ±1, ±2, …
𝑠
The Fourier spectrum 𝑌(𝜔) is shown in figure 2.1.3.1. Now If we have to recover 𝑥(𝑡) from 𝑦(𝑡), we
should be able to recover 𝑋(𝜔) from 𝑌(𝜔), and it is possible only if there is no overlapping between the
successive cycles of 𝑌(𝜔), and for this condition
𝑓𝑠 > 2𝐵 𝐻𝑧
Therefore the sampling interval
1
𝑇𝑠 <
2𝐵

Therefore as long as the sampling frequency 𝑓𝑠 is greater than 2𝐵, 𝑌(𝜔), will consist of non overlapping
repetitions of 𝑋(𝜔), and 𝑥(𝑡) can be recovered from 𝑦(𝑡) by passing 𝑦(𝑡) by an ideal low pass filter with
cut off frequency B Hz.
Nyquist Rate
The minimum sampling rate 𝑓𝑠 = 2𝐵 required to recover 𝑥(𝑡) from its samples 𝑦(𝑡) is called the Nyquist
rate and the corresponding sampling interval is called Nyquist Interval for 𝑦(𝑡).

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Effect of Sampling Rate on 𝒀(𝝎)

Possibility of sampled frequency spectrum with different conditions is given by the following diagrams:

(a) Over Sampling

(b) Under Sampling

(c) Perfect sampling

Figure 2.1.3.2 Sampling Conditions

Aliasing Effect
The overlapped region in case of under sampling represents aliasing effect, which can be removed by
 considering 𝑓𝑠 > 2𝑓𝑚
 By using anti-aliasing filters.

2.2 Sampling of Band Pass Signals

In case of band pass signals, the spectrum of band pass signal X[ω] = 0 for the frequencies outside the
range f1 ≤ f ≤ f2. The frequency f1 is always greater than zero. Plus, there is no aliasing effect when f s > 2f2.
But it has two disadvantages:
 The sampling rate is large in proportion with f2. This has practical limitations.
 The sampled signal spectrum has spectral gaps.
To overcome this, the band pass theorem states that the input signal x(t) can be converted into its samples
and can be recovered back without distortion when sampling frequency 𝑓𝑠 < 2𝑓2.
Also
1 2𝑓2
𝑓𝑠 = =
𝑇 𝑚
Where m is the largest integer < f2 /B and B is the bandwidth of the signal. If f2=KB, then for band pass
signals of bandwidth 2fm and the minimum sampling rate fs=2B=4fm, the spectrum of the sampled signal is
given by
∞
1
𝑌[𝜔] = ∑ 𝑋[𝜔 − 2𝑛𝐵]
𝑇
𝑛=−∞

Aliasing
Aliasing can be referred to as “the phenomenon of a high-frequency component in the spectrum of a
signal, taking on the identity of a low-frequency component in the spectrum of its sampled version.”
The corrective measures taken to reduce the effect of Aliasing are −
 In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the sampler, to
eliminate the high frequency components, which are unwanted.

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The signal which is sampled after filtering is sampled at a rate slightly higher than the Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate, also helps in the easier design of the
reconstruction filter at the receiver.

2.3 Types of Modulation:

In pulse width modulation, there are different types of modulation for analog and digital as shown below:
 PCM: Pulse Code Modulation for Analog Modulation.
 PPM: Pulse Position Modulation for Digital Modulation
 PDM: Pulse Duration Modulation for Digital Modulation.
 PAM: Pulse Amplitude Modulation for Digital Modulation.
Types of Modulation – Tree Diagram:
Types of Modulation

Continuous Wave Modulation Pulse Digital Modulation

Amplitude Modulation Angular Modulation Digital Modulation Analog Modulation

Frequency Modulation Phase Modulation

Pulse Code Modulation

Pulse Amplitude Modulation Pulse Duration Modulation Pulse Position Modulation

Figure 2.3.1 Types of Modulation

PCM is an important method of analog –to-digital conversion. In this modulation

the analog signal is converted into an electrical waveform of two or more levels. A
simple two level waveform is shown in fig 2.3.2.

Figure 2.3.2 A Simple Binary PCM Waveform

The PCM system block diagram is shown in fig 3.2. The essential operations in the transmitter of a PCM
system are Sampling, Quantizing and Coding. The Quantizing and
encoding operations are usually performed by the same circuit, normally referred to as
analog to digital converter. The essential operations in the receiver are regeneration, decoding and

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demodulation of the quantized samples. Regenerative repeaters are used to reconstruct the transmitted
sequence of coded pulses in order to combat the accumulated effects of signal distortion and noise.

2.4 Pulse Amplitude Modulation (PAM):

In pulse amplitude modulation, the amplitude of regular interval of periodic pulses or electromagnetic
pulses is varied in proposition to the sample of modulating signal or message signal. This is an analog type
of modulation. In the pulse amplitude modulation, the message signal is sampled at regular periodic or
time intervals and this each sample is made proportional to the magnitude of the message signal. These
sample pulses can be transmitted directly using wired media or we can use a carrier signal for transmitting
through wireless. There are two types of sampling techniques for transmitting messages using pulse
amplitude modulation, they are
 FLAT TOP PAM: The amplitude of each pulse is directly proportional to instantaneous modulating
signal amplitude at the time of pulse occurrence and then keeps the amplitude of the pulse for the
rest of the half cycle.
 Natural PAM: The amplitude of each pulse is directly proportional to the instantaneous modulating
signal amplitude at the time of pulse occurrence and then follows the amplitude of the modulating
signal for the rest of the half cycle.
Flat top PAM is the best for transmission because we can easily remove the noise and we can also easily
recognize the noise. When we compare the difference between the flat top PAM and natural PAM, flat top
PAM principle of sampling uses sample and hold circuit. In natural principle of sampling, noise interference
is minimum. But in flat top PAM noise interference maximum. Flat top PAM and natural PAM are practical
and sampling rate satisfies the sampling criteria.
There are two types of pulse amplitude modulation based on signal polarity
1. Single polarity pulse amplitude modulation, 2. Double polarity pulse amplitude modulation
In single polarity pulse amplitude modulation, there is fixed level of DC bias added to the message signal or
modulating signal, so the output of modulating signal is always positive. In the double polarity pulse
amplitude modulation, the output of modulating signal will have both positive and negative ends.

Figure 2.4.1 Pulse Amplitude Modulation

Advantages of Pulse Amplitude Modulation (PAM):
 It is the base for all digital modulation techniques and it is simple process for both modulation and
demodulation technique.
 No complex circuitry is required for both transmission and reception. Transmitter and receiver
circuitry is simple and easy to construct.

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 PAM can generate other pulse modulation signals and can carry the message or information at
same time.

Disadvantages of Pulse Amplitude Modulation (PAM):

 Bandwidth should be large for transmitting the pulse amplitude modulation signal. Due to Nyquist
criteria also high bandwidth is required.
 The frequency varies according to the modulating signal or message signal. Due to these variations
in the signal frequency, interferences will be there. So noise will be great. For PAM, noise immunity
is less when compared to other modulation techniques. It is almost equal to amplitude modulation.
 Pulse amplitude signal varies, so power required for transmission will be more, peak power is also,
even at receiving more power is required to receive the pulse amplitude signal.

Applications of Pulse Amplitude Modulation (PAM):

 It is mainly used in Ethernet which is type of computer network communication, we know that we
can use Ethernet for connecting two systems and transfer data between the systems. Pulse
amplitude modulation is used for Ethernet communications.
 It is also used for photo biology which is a study of photosynthesis.
 Used as electronic driver for LED lighting.
 Used in many micro controllers for generating the control signals etc.

2.5 Pulse Position Modulation (PPM):

In the pulse position modulation, the position of each pulse in a signal by taking the reference signal is
varied according to the sample value of message or modulating signal instantaneously. In the pulse
position modulation, width and amplitude is kept constant. It is a technique that uses pulses of the same
breath and height but is displaced in time from some base position according to the amplitude of the signal
at the time of sampling. The position of the pulse is 1:1 which is propositional to the width of the pulse and
also propositional to the instantaneous amplitude of sampled modulating signal. The position of pulse
position modulation is easy when compared to other modulation. It requires pulse width generator and
mono stable multi vibrator.
Pulse width generator is used for generating pulse width modulation signal which will help to trigger the
mono stable multi vibrator; here trial edge of the PWM signal is used for triggering the mono stable multi
vibrator. After triggering the mono stable multi vibrator, PWM signal is converted into pulse position
modulation signal. For demodulation, it requires reference pulse generator, flip-flop and pulse width
modulation demodulator.

Advantages of Pulse Position Modulation (PPM):

 Pulse position modulation has low noise interference when compared to PAM because amplitude
and width of the pulses are made constant during modulation.
 Noise removal and separation is very easy in pulse position modulation.
 Power usage is also very low when compared to other modulations due to constant pulse
amplitude and width.
Disadvantages of Pulse Position Modulation (PPM):
 The synchronization between transmitter and receiver is required, which is not possible for every
time and we need dedicated channel for it.
 Large bandwidth is required for transmission same as pulse amplitude modulation.
 Special equipments are required in this type of modulations.
Applications of Pulse Position Modulation (PPM):
 Used in non-coherent detection where a receiver does not need any Phase lock loop for tracking
the phase of the carrier.
 Used in radio frequency (RF) communication.
 Also used in contactless smart card, high frequency, RFID (radio frequency ID) tags and etc.

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2.6 Pulse Duration Modulation (PDM) or Pulse Width Modulation (PWM):

It is a type of analog modulation. In pulse width modulation or pulse duration modulation, the width of the
pulse carrier is varied in accordance with the sample values of message signal or modulating signal or
modulating voltage. In pulse width modulation, the amplitude is made constant and width of pulse and
position of pulse is made proportional to the amplitude of the signal. We can vary the pulse width in three
ways
1. By keeping the leading edge constant and vary the pulse width with respect to leading edge
2. By keeping the tailing constant.
3. By keeping the center of the pulse constant.
We can generate pulse width using different circuitry. In practical, we use 555 Timer which is the best way
for generating the pulse width modulation signals. By configuring the 555 timer as mono stable or a stable
multi vibrator, we can generate the PWM signals. We can use PIC, 8051, AVR, ARM, etc. microcontrollers
to generate the PWM signals. PWM signal generation has n number of ways. In demodulation, we need
PWM detector and its related circuitry for demodulating the PWM signal.

Advantages of Pulse Width Modulation (PWM):

 As like pulse position modulation, noise interference is less due to amplitude has been made
constant.
 Signal can be separated very easily at demodulation and noise can also be separated easily.
 Synchronization between transmitter and receiver is not required unlike pulse position modulation.
Disadvantages of Pulse Width Modulation (PWM):
 Power will be variable because of varying in width of pulse. Transmitter can handle the power even
for maximum width of the pulse.
 Bandwidth should be large to use in communication, should be huge even when compared to the
pulse amplitude modulation.
Applications of Pulse Width Modulation (PWM):
 PWM is used in telecommunication systems.
 PWM can be used to control the amount of power delivered to a load without incurring the losses.
So, this can be used in power delivering systems.
 Audio effects and amplifications purposes also used.
 PWM signals are used to control the speed of the robot by controlling the motors.
 PWM is also used in robotics.
 Embedded applications.
 Analog and digital applications etc.

PART II DIGITAL TRANSMISSION OF ANALOG SIGNALS

2.7 Quantization
In the process of quantization we create a new signal 𝑚𝑞 (𝑡), which is an approximation to 𝑚(𝑡). The
quantized signal 𝑚𝑞 (𝑡), has the great merit that it is separable from the additive noise.
The operation of quantization is represented in figure 2.7.1. Here we have a signal m(t), whose amplitude
varies in the range from VH to VL as shown in the figure.
We have divided the total range in to M equal intervals each of size S, called the step size and given by
(𝑉𝐻 − 𝑉𝐿 )
𝑆 = ∆=
𝑀
In our example M=8. In the centre of each of this step we located quantization levels 𝑚0, 𝑚1, 𝑚2, … 𝑚7.
The 𝑚𝑞 (𝑡) is generated in the following manner-
Whenever the signal 𝑚(𝑡) is in the range ∆0 , the signal 𝑚𝑞 (𝑡) maintains a constant level 𝑚0 , whenever
the signal 𝑚(𝑡) is in the range ∆1 , the signal 𝑚𝑞 (𝑡) maintains a constant level 𝑚1 and so on. Hence the

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signal 𝑚𝑞 (𝑡) will found all times to one of the levels 𝑚0, 𝑚1, 𝑚2, … 𝑚7. The transition in 𝑚𝑞 (𝑡) from
𝑚0 to 𝑚1 is made abruptly when 𝑚(𝑡) passes the transition level 𝐿01 , which is mid way between 𝑚0
and 𝑚1 and so on.

Figure 2.7.2 Quantization Process

Using quantization of signals, the effect of noise can be reduced significantly. The difference between 𝑚(𝑡)
and 𝑚𝑞 (𝑡) can be regarded as noise and is called quantization noise.
𝑞𝑢𝑎𝑛𝑡𝑖𝑧𝑎𝑡𝑖𝑜𝑛 𝑛𝑜𝑖𝑠𝑒 = 𝑚(𝑡) − 𝑚𝑞 (𝑡)
Also the quantized signal and original signal differs from one another in a ransom manner. This difference
or error due to quantization process is called quantization error and is given by
𝑒 = 𝑚 ( 𝑡 ) − 𝑚𝑘
( )
when 𝑚 𝑡 happens to be close to quantization level 𝑚𝑘 , quantizer output will be 𝑚𝑘 .

The process of transforming sampled amplitude values of a message signal into a discrete amplitude value
is referred to as Quantization. The quantization Process has a two-fold effect:
1. the peak-to-peak range of the input sample values is subdivided into a finite set of decision levels or
decision thresholds that are aligned with the risers of the staircase, and
2. The output is assigned a discrete value selected from a finite set of representation levels that are aligned
with the treads of the staircase.
A quantizer is memory less in that the quantizer output is determined only by the value of a corresponding
input sample, independently of earlier analog samples applied to the input.

Types of Quantizers:
1. Uniform Quantizer
2. Non- Uniform Quantizer
0 Ts 2Ts 3Ts Time Analog Signal Discrete Samples (Quantized)

In Uniform type, the quantization levels are uniformly spaced, whereas in non-uniform type the spacing
between the levels will be unequal and mostly the relation is logarithmic. Types of Uniform Quantizers:
(based on I/P - O/P Characteristics)
1. Mid-Rise type Quantizer

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2. Mid-Tread type Quantizer

In the stair case like graph, the origin lies the middle of the tread portion in Mid –Tread type whereas the
origin lies in the middle of the rise portion in the Mid-Rise type. Mid – tread type: Quantization levels – odd
number. Mid – Rise type: Quantization levels – even number.

Figure 2.7.2 IO Characteristics of Mid-Rise type Quantizer

`
Figure 2.7.3 IO Characteristics of Mid-Tread type Quantizer

2.7.1 Quantization Noise and Signal-to-Noise:

“The Quantization process introduces an error defined as the difference between the input signal, x(t) and
the output signal, y(t). This error is called the Quantization Noise.”

𝑞 (𝑡) = 𝑥(𝑡) − 𝑦(𝑡)

Quantization noise is produced in the transmitter end of a PCM system by rounding off sample values of an
analog base-band signal to the nearest permissible representation levels of the quantizer. As such
quantization noise differs from channel noise in that it is signal dependent.
Let ‘Δ’ be the step size of a quantizer and L be the total number of quantization levels.
Quantization levels are 0, ± Δ., ± 2 Δ., ±3 Δ . . . . . . .
The Quantization error, Q is a random variable and will have its sample values bounded
by [-(Δ/2) < q < (Δ/2)]. If Δ is small, the quantization error can be assumed to a
uniformly distributed random variable.

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Consider a memory less quantizer that is both uniform and symmetric.

L = Number of quantization levels
X = Quantizer input
Y = Quantizer output
The output y is given by
𝑌 = 𝑄(𝑥) …(2.7.1.1)
it is a staircase function that befits the type of mid tread or mid riser quantizer of interest. Suppose that
the input ‘X’ lies inside the interval
𝐼𝑘 = {𝑋𝑘 < 𝑋 ≤ 𝑋𝑘−1 } k=1,2,….L …(2.7.1.2)
Where Xk and Xk−1 are the decision thresholds of the intervals Ik as shown in figure 2.7.1.1.
Correspondingly, the quantizer output y takes on a
discrete value
Y = 𝑦𝑘 if x lies in the interval Ik.
Let q = quantization error with values in the range
∆ ∆
− 2 ≤ 𝑞 ≤ 2 then
Yk = 𝑥 + 𝑞 if ‘n’ lies in the interval Ik
Assuming that the quantizer input ‘n’ is the sample
value of a random variable ‘X’ of zero mean with
variance𝜎𝑥 2 . Figure 2.7.1.1 Decision Thresholds
The quantization noise uniformly distributed out the signal band, its interfering effect on a signal is similar
to that of thermal noise.

2.7.2 Expression for Quantization Noise and SNR in PCM:-

Let Q = Random Variable denotes the Quantization error
q = Sampled value of Q
Assuming that the random variable Q is uniformly distributed over the possible range
(−∆/2 ≤ 𝑞 ≤ ∆/2), as
1 …(2.7.1.3)
𝑓𝑄 (𝑞) = {∆ − ∆/2 ≤ 𝑞 ≤ ∆/2
0 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒

Where fQ(q) = probability density function of the Quantization error. If the signal does not overload the
Quantizer, then the mean of Quantization error is zero and the variance 𝜎𝑥 2 .

Therefore
𝜎𝑄 2 = 𝐸{𝑄2 }
∞
𝜎𝑄 2 = ∫ 𝑞 2 𝑓𝑞 (𝑞 )𝑑𝑞 …(2.7.1.4)
−∞

1 ∆/2 2 ∆2
𝜎𝑄 2 = ∫ 𝑞 𝑑𝑞 = …(2.7.1.5)
∆ −∆/2 12

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Thus the varience of the Quantization noise produced by a Uniform Quantizer, grows as the square of the
step size. Equation (2.7.1.5) gives an expression for Quantization noise in PCM system.
Let 𝜎𝑥 2 = Variance of the base band signal x(t) at the input of the quantizer.
When the base band signal is reconstructed at the receiver output, we obtain original signal plus
Quantization noise. Therefore output signal to Quantization noise ration (SNR) is given by

𝑆𝑖𝑔𝑛𝑎𝑙 𝑃𝑜𝑤𝑒𝑟 𝜎𝑥 2 𝜎𝑥 2
(𝑆𝑁𝑅)𝑄 = = 2= 2 …(2.7.1.6)
𝑁𝑜𝑖𝑠𝑒 𝑃𝑜𝑤𝑒𝑟 𝜎𝑄 ∆ /12
Smaller the step size ∆, larger will be the SNR.

2.7.3 Signal to Quantization Noise Ration:- [Mid Tread Type]

Let x = Quantizer input, sampled value of random variable X with mean X, variance 𝜎𝑥 2. The Quantizer is
assumed to be uniform, symmetric and mid trade type.
Xmax=absolute value of the overload level of the Quantizer.
∆= Step size,
Then L= No. of Quantization level given by
2𝑋𝑚𝑎𝑥
𝐿= +1 …(2.7.1.7)
∆
Let n = no. of bits used to represent each level.
In general 2n = L, but in the mid trade quantizer, since the number of representation levels is odd,
𝐿 = 2𝑛 − 1
…(2.7.1.8)

From equations (2.7.1.7) and (2.7.1.8),

2𝑋𝑚𝑎𝑥
2𝑛 − 1 = +1
∆
Or
𝑋𝑚𝑎𝑥
∆= 𝑛−1
…(2.7.1.9)
2 − 1
𝑋𝑚𝑎𝑥
The ration is called the loading factor. To avoid significant overload distortion, the amplitude of the
𝜎𝑥
Quantizer input x extend from −4𝜎𝑥 to 4𝜎𝑥 , which correspond to loading factor of 4. Thus with 𝑋𝑚𝑎𝑥 =
4𝜎𝑥 , we can write equation (2.7.1.9) as,
4𝜎𝑥
∆= 𝑛−1 …(2.7.1.10)
2 −1
𝜎𝑥 2 3
(𝑆𝑁𝑅)𝑄 = 2 = [2𝑛−1 − 1]2 …(2.7.1.11)
∆ /12 4
For larger value of n(typically n>6), we may approximate the result as
3 3 2𝑛
(𝑆𝑁𝑅)𝑄 = [2𝑛−1 − 1]2 ≈ (2 ) …(2.7.1.12)
4 16
Hence expressing the SNR in db
10𝑙𝑜𝑔10 (𝑆𝑁𝑅)𝑄 = 6𝑛 − 7.2
…(2.7.1.13)
This formula states that each bit in code word of a PCM system contributes 6db to the
signal to noise ratio. For loading factor of 4, the problem of overload i.e. the problem that the sampled
value of signal falls outside the total amplitude range of Quantizer, 8σx is less than 10-4.
The equation 2.7.1.11 gives a good description of the noise performance of a PCM
system provided that the following conditions are satisfied.
1. The Quantization error is uniformly distributed

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2. The system operates with an average signal power above the error threshold so
that the effect of channel noise is made negligible and performance is there by
limited essentially by Quantization noise alone.
3. The Quantization is fine enough (say n>6) to prevent signal correlated patterns in
the Quantization error waveform
4. The Quantizer is aligned with input for a loading factor of 4
Note: 1. Error uniformly distributed
2. Average signal power
3. n > 6
4. Loading factor = 4
From (2.7.1.13): 10 log10 (SNR)O = 6n – 7.2

In a PCM system, Bandwidth B = nW or [n=B/W], substituting the value of ‘n’ we get

𝐵
10𝑙𝑜𝑔10 (𝑆𝑁𝑅)𝑄 = 6( ) − 7.2 …(2.7.1.14)
𝑊

2.7.4 Signal to Quantization Noise Ratio:- [Mid Rise Type]

Let x = Quantizer input, sampled value of random variable X with mean X, variance 𝜎𝑥 2. The Quantizer is
assumed to be uniform, symmetric and mid rise type.
Let Xmax=absolute value of the overload level of the Quantizer.
2𝑋𝑚𝑎𝑥
𝐿= …(2.7.1.15)
∆
Since the number of representation levels is odd,
𝐿 = 2𝑛 (Mid rise only) …(2.7.1.16)

From equations (2.7.1.15) and (2.7.1.16),

𝑋𝑚𝑎𝑥
∆= …(2.7.1.17)
2𝑛
𝜎𝑥 2
(𝑆𝑁𝑅)𝑄 = 2 …(2.7.1.18)
∆ /12
Where 𝜎𝑥 2 represent the variance or the signal power.

Consider a special case of sinusoidal signals:

2
Let the signal power be Ps, then 𝑃𝑠 = 0.5𝑋𝑚𝑎𝑥
𝑃𝑠 12𝑃𝑠
(𝑆𝑁𝑅)𝑄 = 2
= = 1.5𝐿2 = 1.5𝐿2𝑛 …(2.7.1.19)
∆ /12 ∆2
In decibels (𝑆𝑁𝑅)𝑄 = 1.76 + 6.02𝑛
…(3.20)
Improvement of SNR can be achieved by increasing the number of bits, n. Thus for ‘n’ number of bits/
sample the SNR is given by the above equation 2.7.1.19. For every increase of one bit / sample the step
size reduces by half. Thus for (n+1) bits the SNR is given by
(𝑆𝑁𝑅)(𝑛+1)𝑏𝑖𝑡 = (𝑆𝑁𝑅)(𝑛)𝑏𝑖𝑡 + 6 𝑑𝐵

Therefore addition of each bit increases the SNR by 6dB.

2.7.5 Classification of Quantization Noise:

The Quantizing noise at the output of the PCM decoder can be categorized into four types depending on
the operating conditions, Overload noise, Random noise, Granular Noise and Hunting noise
Over Load Noise:- The level of the analog waveform at the input of the PCM encoder needs to be set so
that its peak value does not exceed the design peak of V max volts. If the peak input does exceed Vmax, then

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the recovered analog waveform at the output of the PCM system will have flat – top near the peak values.
This produces overload noise.
Granular Noise:- If the input level is reduced to a relatively small value w.r.to to the design level
(quantization level), the error values are not same from sample to sample and the noise has a harsh sound
resembling gravel being poured into a barrel. This is granular noise. This noise can be randomized (noise
power decreased) by increasing the number of quantization levels i.e. Increasing the PCM bit rate.
Hunting Noise:- This occurs when the input analog waveform is nearly constant. For these conditions, the
sample values at the Quantizer output can oscillate between two adjacent quantization levels, causing an
undesired sinusoidal type tone of frequency (0.5fs) at the output of the PCM system. This noise can be
reduced by designing the quantizer so that there is no vertical step at constant value of the inputs.

2.7.6 Quantization Error

For any system, during its functioning, there is always a difference in the values of its input and output. The
processing of the system results in an error, which is the difference of those values.
The difference between an input value and its quantized value is called a Quantization Error. A Quantizer is
a logarithmic function that performs Quantization (rounding off the value). An analog-to-digital converter
(ADC) works as a quantizer.
The following figure illustrates an example for a quantization error, indicating the difference between the
original signal and the quantized signal.

Figure 2.7.6.1 Quantization Error

Quantization Noise
It is a type of quantization error, which usually occurs in analog audio signal, while quantizing it to digital.
For example, in music, the signals keep changing continuously, where regularity is not found in errors. Such
errors create a wideband noise called as Quantization Noise.

2.8 Pulse Code Modulation:

A signal which is to be quantized before transmission is sampled as well. The quantization is used to reduce
the effect of noise and the sampling allows us to do the time division multiplexing. The combined
operation of sampling and quantization generate a quantized PAM waveform i.e. a train of pulses whose
amplitude is restricted to a number of discrete levels.
Rather than transmitting the sampled values itself, we may represent each quantization level by a code
number and transmit the code number. Most frequently the code number is converted in to binary

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equivalent before transmission. Then the digits of the binary representation of the code are transmitted as
pulses. This system of transmission is called binary Pulse Code Modulation. The whole process can be
understood by the following diagram.

Figure 2.8.1 Pulse Code Modulation

PCM Transmitter:
Basic Blocks:
1. Anti aliasing Filter, 2. Sampler, 3. Quantizer, 4. Encoder
The block diagram of a PCM transmitter is shown in figure (a). An anti-aliasing filter is basically a filter used
to ensure that the input signal to sampler is free from the unwanted frequency components. For most of
the applications these are low-pass filters. It removes the frequency components of the signal which are
above the cutoff frequency of the filter. The cutoff frequency of the filter is chosen such it is very close to
the highest frequency component of the signal.
The message signal is sampled at the Nyquist rate by the sampler. The sampled pulses are then quantized
by the quantizer. The encoder encodes these quantized pulses in to binary equivalent, which are then
transmitted over the channel. During the channel the regenerative repeaters are used to maintain the
signal to noise ratio.
Continuous time
PCM Wave
message signal

LPF Sampler Quantizer Encoder

(A) Transmitter

Distorted Regenerative Regenerative

PCM wave Repeater Repeater

(b) Transmission Path

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Input
Message
Quantizer Holding Circuit LPF
Decoder

(c) Receiver

Figure 2.8.2 PCM System Basic Block Diagram

Figure (c) shows the receiver. The first block is again the quantizer, but this quantizer is different from the
transmitter quantizer sa it has to take the decision regarding the presence or absence of the pulse only.
Thus there are only two quantization levels. The output of the quantizer goes to the decoder which is an
D/A converter that performs the inverse operation of the encoder. The decoder output is a sequence of
quantized pulses. The original signal is reconstructed in the holding circuit and the LPF.

Figure 2.8.3 PCM Encoding

2.8.1 Advantages of Pulse Code Modulation:

 Pulse code modulation will have low noise addition and data loss is also very low.
 We can repeat the exact transmitted signal at the receiver. This is called repeatability. And we can
retransmit the signal with any distortion loss also.
 Pulse code modulation is used in music play back CD’s and also used in DVD for data storing whose
sampling rate is bit higher.
 Pulse code modulation can be used in storing the data.
 PCM can encode the data also.
 Multiplexing of signals can also be done using pulse code modulation. Multiplexing is nothing for
adding the different signals and transmitting the signal at same time.
 Pulse code modulation requires large bandwidth
 Pulse code modulation permits the use of pulse regeneration.

2.8.2 Disadvantages of Pulse Code Modulation:

 Specialized circuitry is required for transmitting and also for quantizing the samples at same
quantized levels. We can do encoding using pulse code modulation but we need to have complex
and special circuitry.

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 Pulse code modulation receivers are cost effective when we compared to other modulation
receivers.
 Developing pulse code modulation is bit complicated and checking the transmission quality is also
difficult and takes more time.
 Large bandwidth is required for pulse code modulation when compared to bandwidth used by the
normal analog signals to transmit message.
 Channel bandwidth should be more for digital encoding.
 PCM systems are complicated when compared to analog modulation methods and other systems.
 Decoding also needs special equipment’s and they are also too complex.

2.8.3 Applications of Pulse Code Modulation (PCM):

 Pulse code modulation is used in telecommunication systems, air traffic control systems etc.
 Pulse code modulation is used in compressing the data that is why it is used in storing data in
optical disks like DVD, CDs etc. PCM is even used in the database management systems.
 Pulse code modulation is used in mobile phones, normal telephones etc.
Remote controlled cars, planes, trains use pulse code modulations.

2.9 Companding in PCM

The word Companding is a combination of Compressing and Expanding, which means that it does both.
This is a non-linear technique used in PCM which compresses the data at the transmitter and expands the
same data at the receiver. The effects of noise and crosstalk are reduced by using this technique.
There are two types of Companding techniques. They are −
A-law Companding Technique
 Uniform quantization is achieved at A = 1, where the characteristic curve is linear and no
compression is done.
 A-law has mid-rise at the origin. Hence, it contains a non-zero value.
 A-law companding is used for PCM telephone systems.

2.9.1 µ-law Companding Technique

 Uniform quantization is achieved at µ = 0, where the characteristic curve is linear and no
compression is done.
 µ-law has mid-tread at the origin. Hence, it contains a zero value.
 µ-law companding is used for speech and music signals.
µ-law is used in North America and Japan.
For the samples that are highly correlated, when encoded by PCM technique, leave redundant information
behind. To process this redundant information and to have a better output, it is a wise decision to take a
predicted sampled value, assumed from its previous output and summarize them with the quantized
values. Such a process is called as Differential PCM (DPCM) technique.

2.10 Differential Pulse Code Modulation:

Differential Pulse Code Modulation is an alternative to PCM. Instead of transmitting the sampled values
itself at each sampling time; we can transmit the difference between the two successive samples. For
example, we can transmit the difference between the sample value 𝑚(𝑘) at sampling time K and sample
value 𝑚(𝑘 − 1) at sampling time k-1. If such changes are transmitted then at the receiving end we can
generate a waveform identical to the m(t) by simply adding up these changes.
The DPCM has the special merit that when these differences are transmitted by PCM. The differences
𝑚(𝑘) − 𝑚(𝑘 − 1) will be smaller than the sample values themselves and fewer levels will be required to

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quantize 𝑚(𝑘), and corresponding fewer bits will be needed to encode the signal. The basic principle of
DPCM is shown in figure 2.10.1.

(a) Transmitter

(b) Receiver
Figure 2.10.1 Differential PCM

The receiver consists of an accumulator which adds-up the receiver quantized differences ∆𝑄 (𝑘) and a
filter which smoothes out the quantization noise. The output of accumulator is the signal approximation
𝑚̂ (𝑘) which becomes 𝑚 ̂ (𝑡) at the filter output.
At the transmitter we need to know whether the 𝑚 ̂ (𝑡) is larger or smaller than 𝑚(𝑡) and by how much
amount. We may than determine whether the next difference ∆𝑄 (𝑘) needs to be positive or negative and
of what amplitude in order to bring 𝑚 ̂ (𝑡) as close as possible to 𝑚(𝑡). For this reason we have a duplicate
accumulator at transmitter.
At each sampling time the transmitter difference amplifier compares 𝑚(𝑡) and ̂(𝑡), 𝑚 and the sample and
hold circuit holds the result of that comparison ∆(𝑡), for the duration of interval between sampling times.
The quantizer generates the signal 𝑆0 (𝑡) = ∆𝑄 (𝑘) both for the transmission to the receiver and to provide
the input to the receiver accumulator in the transmitter.
The basic limitation of the DPCM scheme is that the transmitted differences are quantized and are of
limited values.
Need for a predictor:
There is a correlation between the successive samples of the signal 𝑚(𝑡). To take the advantage of this
correlation a predictor is included. It needs to incorporate the facility for storing past differences and
carrying out some algorithm to predict then next required increment.

2.11 Delta Modulation:

Delta Modulation is a DPCM scheme in which the difference signal ∆(𝑡) is encoded into just a single bit.
The single bit providing just for two possibilities is used to increase or decrease the estimate ̂𝑚(𝑡)[𝑚𝑞 (𝑡)].
The Linear Delta Modulator is shown in figure 2.11.1.
The baseband signal 𝑚(𝑡) and its quantized approximation ̂ 𝑚(𝑡) are applied as input to a comparator. The
comparator has one fixed output V(H) when 𝑚(𝑡) > 𝑚𝑞 (𝑡) and a difference output V(L) when 𝑚(𝑡) <
𝑚𝑞 (𝑡). Ideally the transition between V(H) and V(L) is arbitrarily abrupt as 𝑚(𝑡) − 𝑚𝑞 (𝑡) passes through

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zero. The up-down counter increments or

decrements its count by 1 at each active edge of
the clock waveform. The count direction i.e.
incrementing or decrementing is determined by
the voltage levels at the “Count direction
command” input to the counter. When this binary
input is at level V(H), the counter counts up and
when this binary input is at level V(L), the counter
counts down.
The digital output of the counter is converted into
analog quantized approximation 𝑚𝑞 (𝑡) by a D/A
converter. The waveforms for the delta
modulator of figure 2.11.1 is shown in figure
2.11.2, assuming that the active clock edge is
falling edge.
It may be noted that at startup there is a brief
interval when 𝑚𝑞 (𝑡) may be a poor
approximation to 𝑚(𝑡), as shown in figure 2.11.3.
The initial large discrepancy between 𝑚(𝑡) and
𝑚𝑞 (𝑡) and stepwise approach of 𝑚𝑞 (𝑡) to 𝑚(𝑡) is
Figure 2.11.1 Delta Modulator
shown in figure 2.11.3.

Figure 2.11.2 The response of the delta modulator to a baseband signal m(t)

Figure 2.11.3 Startup response of DM

Figure 2.11.4 Slope Overload in a linear DM

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It should be noted that when 𝑚𝑞 (𝑡) has caught up 𝑚(𝑡) and even though 𝑚(𝑡) remains constant, 𝑚𝑞 (𝑡)
hunts, swinging up and down to 𝑚(𝑡).
Slope Overload
The excessive disparity between 𝑚(𝑡) and 𝑚𝑞 (𝑡) is described as a slope overload error and occurs
whenever 𝑚(𝑡) has a slope larger than the slope 𝑆/𝑇𝑠 which can be sustained by the waveform 𝑚𝑞 (𝑡). The
slope overload as shown in figure 3.11.4 is developed due to the small size of S. To overcome the overload
we have to increase the sampling rate above the rate initially selected to satisfy the Nyquist criterion. The
sampling rate 𝑓𝑠 must satisfy the following condition
𝑠𝑓𝑠 = 2𝜋𝑓𝐴

Features of DM
Following are some of the features of delta modulation.
 An over-sampled input is taken to make full use of the signal correlation.
 The quantization design is simple.
 The input sequence is much higher than the Nyquist rate.
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 The stair-case approximation of output waveform.
 The step-size is very small, i.e., Δ (delta).
 The bit rate can be decided by the user.
 This involves simpler implementation.

2.11.3 Advantages of DM Over DPCM

 1-bit quantizer
 Very easy design of the modulator and the demodulator
 However, there exists some noise in DM.
 Slope Over load distortion (when Δ is small)
 Granular noise (when Δ is large)

2.12 Adaptive Delta Modulation (ADM):

In digital modulation, we have come across certain problem of determining the step-size, which influences
the quality of the output wave.
A larger step-size is needed in the step slope of modulating signal and a smaller step size is needed where
the message has a small slope. The minute details get missed in the process. So, it would be better if we
can control the adjustment of step-size, according to our requirement in order to obtain the sampling in a
desired fashion. This is the concept of Adaptive Delta Modulation.
Following is the block diagram of Adaptive delta modulator.

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Figure 2.12.1 Adaptive Delta Modulation (ADM)

The step size S is not of fixed size but it is always a multiple of basic step size 𝑆0 . The basic step size 𝑆0 is
either added or subtracted by the accumulator as required to move 𝑚𝑞 (𝑡) more close to 𝑚(𝑡). If the
direction of the step at the clock edge K is same as at edge K-1, then the processor increases the step size
by an amount 𝑆0 . If the directions are opposite then the processor decreases the magnitude of the step
by 𝑆0 .

Figure 2.12.2 Waveforms comparing the response of DM and ADM

In figure 2.12.1, the output 𝑆0 (𝑡) is called 𝑒(𝑘), which represents the error i.e. the discrepancy between
the 𝑚(𝑡) and 𝑚𝑞 (𝑡), and it is either V(H) or V(L).

𝑒(𝑘) = +1, 𝑖𝑓 𝑚(𝑡) > 𝑚𝑞 (𝑡) immediately before Kth edge,

𝑒(𝑘) = −1, 𝑖𝑓 𝑚(𝑡) < 𝑚𝑞 (𝑡) immediately before Kth edge,

The features of ADM are shown in figure 2.12.2. As long as the condition 𝑚(𝑡) > 𝑚𝑞 (𝑡) persists the jumps
in 𝑚𝑞 (𝑡) becomes larger, that’s why 𝑚𝑞 (𝑡) catches up with 𝑚(𝑡) sooner than in the case of linear DM, as
shown by 𝑚′𝑞 (𝑡).
On the other hand, when the response to the large slope in 𝑚(𝑡), 𝑚𝑞 (𝑡) develops large jumps and large
number of clock cycles are required for these jumps to settle down. Therefore the ADM system reduces
the slope overload but it increases the quantization error. Also when 𝑚(𝑡) is constant 𝑚𝑞 (𝑡) oscillates
about 𝑚(𝑡) but the oscillation frequency is half of the clock frequency.

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2.13 Voice coders:

A vocoder i.e. voice encoder is an analysis/synthesis system, used to reproduce human speech. In the
encoder, the input is passed through a multiband filter, each band is passed through an envelope follower,
and the control signals from the envelope followers are communicated to the decoder. The decoder
applies these control signals to corresponding filters in the (re)synthesizer.

It was originally developed as a speech coder for telecommunications applications in the 1930s, the idea
being to code speech for transmission. Its primary use in this fashion is for secure radio communication,
where voice has to be encrypted and then transmitted. The advantage of this method of "encryption" is
that no 'signal' is sent, but rather envelopes of the band pass filters. The receiving unit needs to be set up
in the same channel configuration.
Information, and recreates it, The Voder i.e. Voice Operating Demonstrator generates synthesized speech
by means of a console with fifteen touch-sensitive keys and a pedal, basically consisting of the "second
half" of the vocoder, but with manual filter controls, needing a highly trained operator.

The human voice consists of sounds generated by the opening and closing of the glottis by the vocal cords,
which produces a periodic waveform with many harmonics. This basic sound is then filtered by the nose
and throat (a complicated resonant piping system) to produce differences in harmonic content (formants)
in a controlled way, creating the wide variety of sounds used in speech. There is another set of sounds,
known as the unvoiced and plosive sounds, which are created or modified by the mouth in different
fashions.
The vocoder examines speech by measuring how its spectral characteristics change over time. This results
in a series of numbers representing these modified frequencies at any particular time as the user speaks.
In simple terms, the signal is split into a number of frequency bands (the larger this number, the more
accurate the analysis) and the level of signal present at each frequency band gives the instantaneous
representation of the spectral energy content. Thus, the vocoder dramatically reduces the amount of
information needed to store speech, from a complete recording to a series of numbers. To recreate
speech, the vocoder simply reverses the process, processing a broadband noise source by passing it
through a stage that filters the frequency content based on the originally recorded series of numbers.
Information about the instantaneous frequency (as distinct from spectral characteristic) of the original
voice signal is discarded; it wasn't important to preserve this for the purposes of the vocoder's original use
as an encryption aid, and it is this "dehumanizing" quality of the vocoding process that has made it useful
in creating special voice effects in popular music and audio entertainment.

Since the vocoder process sends only the parameters of the vocal model over the communication link,
instead of a point by point recreation of the waveform, it allows a significant reduction in the bandwidth
required to transmit speech.

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