Module – 5 Digital Representation of Analog Signals 15EC45

5.1 Introduction
Converting a continuous analog signal into a digital signal has two sub-processes:
1. Sampling - conversion of a continuous-space/time(audio, video) signal into a discrete-space/time
(audio, video) signal.
2. Quantization - converting a continuous-valued (audio, video) signal that has a continuous range (set
of values that it can take) of intensities and/or colors into a discrete-valued (audio, video) signal that has
a discrete range of intensities and/or colors; this is usually done by rounding, truncation or other
irreversible non-linear process of information destruction
In the first step from analog to digital, an analog source is sampled at discrete times. The resulting
analog samples are then transmitted by means of analog pulse modulation (Ex: PAM).

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In the second step from analog to digital, an analog source is not only sampled at discrete times but
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also the samples themselves are also quantized to discrete levels.

Pulse-code Modulation (PCM)

Delta Modulation (DM)
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Advantages of digital transmission over analog transmission

1. Digital systems are less sensitive to noise than analog.

For long transmission lengths, the signal may be regenerated effectively error-free at different
point along the path and the original signal transmitted over the remaining length.
2. With digital systems, it is easier to integrate different services.
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Example: video and the accompanying soundtrack, into the same transmission scheme.
3. The transmission scheme can be relatively independent of the source.
Example: a digital transmission scheme that transmits voice at 10 kbps could also be used to
transmit computer data at 10 kbps.
4. Circuitry for handling digital signals is easier to repeat and digital circuits are less sensitive to
physical effect such as vibration and temperature.
5. Digital signals are simpler to characterize and typically do not have the same amplitude range and
variability as analog signals.
This makes the associated hardware easier to design.
Digital techniques offer strategies for more efficient use of media, example, cable, radio wave, and
optical fibers.
1. Various media sharing strategies, known as multiplexing techniques, are more easily implemented
with digital transmission strategies.

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2. There are techniques for removing redundancy from a digital transmission, so as to minimize the
amount of information that has to be transmitted. These techniques fall under the broad
classification of source coding
3. There are techniques for adding controlled redundancy to digital transmission, such that errors
occur during transmission may be corrected at the receiver without any additional information.
These techniques fall under the general. These techniques fall under the general category of
channel coding.
4. Digital techniques make it easier to specify complex standards that may be shared on a worldwide
basis. This allows the development of communication components with many different features
(e.g., a cellular handset) and their interoperation with a different component (e.g., a base station)
produced by a different manufacturer.

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5. Other channel compensations techniques, such as equalization, especially adaptive versions, are
easier to implement with digital transmission techniques.

2. The Sampling Process

Sampling process is used to convert a continuous-time signal into a discrete-time sequence. DT

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signal, g(n) is obtained by extracting CT signal, g(t) every Ts seconds, where Ts is known as the
sampling period or interval.
For lossless digitization, the sampling rate should be at least twice the maximum frequency response.
In mathematical terms:
fs > 2*fm
Where fs is sampling frequency and fm is the maximum frequency in the signal.
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Fig. 1 .
Sampling process

In time domain, gδ(t) = g(t) · p(t), where p(t) is a periodic impulse train defined as

p (t )   (t  nT )
s
..................... (5.1)
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n 


 (t  nT )
Then, gδ(t) = g(t) · n 
s
since, t = nTs and using sifting property of impuse,

g  (t )   g (nT )(t  nT )
n 
s s
..................... (5.2)
Frequency domain represntation of sampled signal of Eqn (5.2) is obtained by applying CT Fourier
Transform on both sides,
  
FT [ g  (t )]  FT   g (nTs ) (t  nTs ) 
 n  
Multiplication in time domain = convolution
in frequency domain
1  

G ( f )   G ( f )  2 f s   ( f  nf s ) 
2  n  
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.......... (5.3)

Fig. 2 Spectrum of band limited signal

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Expansion of this eqn. is given below and the same is shown in the fig. 2(b)

G ( f )  .....  f s G ( f  2 f s )  f s G ( f  f s )  f s G ( f )  f s G ( f  f s )  f s G ( f  2 f s )  .....

Using the following two conditions,

Then,
6DTFT of G(f)→G(Ω)
G ( f )  f s G ( f )


or RA G( f ) 
G ( f )
fs ..................... (5.4)
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G ()   g ( n )e
n 
 j 2 fn

for n = nTs

G ( f )   g (nT )e  j 2 f ( nTs )
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s
n  ..................... (5.5)
Substitute Eq (5.5) in Eq (5.4)

1
G( f ) 
fs
 g (nT )e s
 j 2 f ( nTs )

n  ............... (5.6)

Taking inverse FT for Eqn (5.6)
1 

F 1 G ( f )   F 1   g (nT )e s
 j 2 f ( nTs )

 fs 
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n 


1  
  f s n
g (t )  g (nTs )e  j 2 f ( nTs ) e j 2 ft df

 ;-W≤ f≤W
Interchanging order of summation and integration
 W
1
 
n 
g (nTs ) 
f s W
e j 2 f (t  nTs ) df


1  e j 2W (t  nTs )  e  j 2W (t  nTs )  e j  e  j
  g (nTs )  j 2 ( t  nTs )  sin   j2
n  fs   since,

1  e j 2W (t  nTs )  e  j 2W (t  nTs ) 
  g (nTs ) 
fs  ( t  nTs ) 
j2 
n  

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  sin(2 W ( t  nT )) 
  g (nT )
n 
s



s 
f s ( t  nTs ) 


 sin(2 W (t  nTs ))  1 1
 g (nT ) 
s
f s (t  nTs ) 
 Ts  
2W f s
= n   since,

.................... (5.7)
That means sequence {g(n/2W)}has all the information contained in g(t).

Reconstructing the signal of g(t): Comments From Eq. (5.7):

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1. Provides an interpolation formula for reconstructing the original signal from the
sequence of sample values {g(n/2W)}
2. sinc(2Wt) playing the role of an interpolation function
3. Eqn (5.7) represents the convolution (or filtering) of the impulse train gδ(t) given by Eq. (5.2) with
the impulse response sinc(2Wt).
Any impulse response that plays the same roles as sinc(2Wt) is also referred to as a reconstruction

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filter. The sampling theorem for strictly band-limited signals of finite energy may be stated in
two equivalent parts. A band-limited signal of finite energy, which only has frequency
components less than W Hz.
1. It is completely described by specifying the values of the signal at instants of time
separated by 1/2W seconds.
2. It may be, completely recovered from a knowledge of its samples taken at the rate of 2W
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samples per second.
The sampling rate of 2W samples per second, for a signal bandwidth of W Hz, is called the
Nyquist rate; its reciprocal 1/2W (measured in seconds) is called Nyquist interval.
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The reconstruction filter is low-

pass with a passband extending
from –W to W

The reconstruction filter has a

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transition band extending from

W to fs-W.

The fact that the reconstruction

filter has a well defined
transition band, means
that it is physically realizable

Fig. 3 Reconstructing the signal of g(t)

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Corrective measures to avoid aliasing effects:

 Prior to sampling, a LPF is used as pre-alias filter to attenuate those high frequency
components of the signal that are not essential to the information components of the
signal.
 The filtered signal is sampled at a rate slightly higher than the Nyquist rate
Typical bandwidths and sampling frequencies in signal processing applications

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3. Pulse Amplitude Modulation
In pulse-amplitude modulation (PAM), the amplitudes of regularly spaced pulses are varied in

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proportion to the corresponding sample values of a continuous message signal; the pulses can be of a
rectangular form or some other shape. PAM is somewhat similar to natural sampling, where the
message signal is multiplied by a periodic train of rectangular pulses. In natural sampling the top of
each modulated rectangular pulse varies with the message signal.

Two steps are involved in the generation of the PAM signal.

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1. Instantaneous sampling of the message signal m(t) every Ts seconds, where the sampling rate
fs = 1/Ts is chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample so obtained to some constant value T.
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In digital circuit technology, these two operations are jointly referred to as “sample and hold.”
One important reason for intentionally lengthening the duration of each sample is to avoid the use of
an excessive channel bandwidth, since bandwidth is inversely proportional to pulse duration.
One important reason for intentionally
lengthening the duration T of each sample
is to avoid the use of an excessive channel
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bandwidth, since bandwidth is inversely

proportional to pulse duration.
The amplitude of each pulse is directly
proportional to modulating signal amplitude at
the time of pulse occurrence. The tops of the
amplitude remain flat.
Fig.4 Sampled waveform with flat top pulses
Sample Hold Analysis: Let s(t) denote the
sequence of flat-top pulses generated
when m(t) is convolved with train of h(t),
which is a standard rectangular pulse of
unit amplitude and duration T, defined as

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Hence, we may express the PAM signal as

s(t) = mδ(t) * h(t), and is defined as, for t =

nTs

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Where Ts is the sampling period and
m(nTs) is the sample value of obtained at
time t = nTs.

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Fig.5 standard rectangular pulse of unit amplitude
and duration T.

By definition, the instantaneously sampled version of m(t) is given by

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.................... (5.8)
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where δ(t-nTs) is a time-shifted delta function.

To convert mδ(t) into the PAM signal s(t) we convolve mδ(t) with the pulse h(t) obtaining
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.................... (5.9)

where, in the last line, we have interchanged the order of summation and integration, both
of which are linear operations. Using the sifting property of the delta function,

.................... (5.10)

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Therefore, the PAM signal is mathematically equivalent to the convolution of mδ(t) with the
pulse h(t)

Taking the Fourier transform on both sides,

FT

Therefore, substitution of Mδ(f ) in the above equation,

.................... (5.11)

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Note that, using flat-top samples to generate a PAM signal, we have introduced amplitude
distortion as well as a delay of T/2 (known as aperture effect).

This distortion may be corrected by connecting an equalizer in cascade with the low-pass
reconstruction filter.

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The equalizer has the effect of decreasing the in-band loss of the reconstruction filter.
Ideally, the amplitude response of the equalizer is given by,
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.................... (5.12)
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PAM signal imposes i) stringent requirements on the amplitude and phase responses of the
channel, because of the relatively short duration of the transmitted pulses ii) the noise
performance of a PAM system can never be better than direct transmission of the message
signal. Accordingly, we find that for transmission over long distances, PAM would be used
only as a means of message processing for time-division multiplexing.

5.4 Time Division Multiplexing (TDM)

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TDM is a method of transmitting and receiving multiple independent message signals over a single
transmission channel. Transmission of the message samples engages the communication channel for
only a fraction of the sampling interval on a periodic basis.
Each message signal is first restricted in bandwidth by a low pass pre-alias filter to remove the
frequencies that are not essential which helps in reducing the aliasing problem. The outputs of these
filters are then applied to a commutator.
Electronic commutator
i) sequentially samples each of the N input messages at a rate fs > 2W, where W is the cut-off
frequency of the pre-alias filter
ii) Combines them to form a composite base band signal, which travels through the channel.
The multiplexed signal is then applied to a pulse amplitude modulator.

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Pulse modulator: This transforms the multiplexed signal into a form suitable for transmission over the
communication channel.
At the receiver end of the system, the received signal is applied to a pulse amplitude demodulator,
which performs the reverse operation of the pulse amplitude modulator.

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Fig.6 Block Diagram of Time Division Multiplexing

Pulse demodulator: Performs the reverse operation of the pulse modulator.

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Decommutator: The narrow samples produced at the pulse demodulator output are distributed
to the appropriate low-pass reconstruction filters. The decommutator operates in synchronism with
the commutator in the transmitter.

5.5 Pulse Position Modulation

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PPM is a modulation technique designed to achieve the goals like simple transmitter and receiver
circuitry, noise performance, constant bandwidth - efficiency and constant transmitter power. In PPM
the amplitude and width of the pulses is kept constant but the position of each pulse is varied accordance
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with the amplitudes of the sampled value of the modulating signal. The position of the pulses is changed
with respect to the position of reference pulses.
The PPM pulses can be derived from the PWM pulses. Each trailing of the pulse width modulated
signal becomes the starting point for pulses in PPM signal.
Pulse-Duration Modulation (PDM): This form of modulation is also referred to as Pulse-Width
Modulation. PWM is sometimes called pulse duration modulation (PDM) or pulse length modulation
(PLM), as the width (active portion of the duty cycle) of a constant amplitude pulse is varied
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proportional to the amplitude of the message signal at the time the signal is sampled.
The maximum analog signal amplitude produces the widest pulse, and the minimum analog signal
amplitude produces the narrowest pulse. Note, however, that all pulses have the same amplitude.
Mathematical representation of PPM signal using the sample m(nTs) of a message signal m(t) to
modulate the position of nth pulse, is

................ (5.13)
Where, kp is sensitivity constant of PPM. m(t) denotes a standard pulse of interest. The different
pulses constituting the PPM signal s(t) must be strictly non-overlapping for this a sufficient condition
is given by:

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................ (5.14)

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Fig. 7 Block diagram for generating PPM waves
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5.5.1 Generation of PPM waves
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The message signal m(t) is first

converted in to a PAM signal by means
of a sample-and-hold circuit, generating
a staircase waveform u(t).

Next, the signal u(t) is added to a

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sawtooth wave, yielding the combined

signal v(t)
The v(t) is applied to a threshold
detector that produces a very narrow
pulse (approximating an impulse)
each time v(t) crosses zero in
the negative-going direction.

Finally the PPM signal s(t) is

generated by using this sequence of
impulses to excite a filter whose impulse
response is defined by the
standard pulse g(t). See fig. 9.

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Fig.8 Gereration of PPM signal

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Fig. 9 Block diagram for generating PPM waves
1.

Detection of PPM wave

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Operation of linear type of PPM receiver is as follows:

1. Convert PPM into PDM using edge triggered FFs.

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2. This PDM wave is integrated with finite time by computing the area of each pulse.
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3. Sample the output of integrator at a uniform rate, where pulse amplitudes are proportional to
samples m(nTs) of original PPM wave s(t).
4. Finally demodulate PAM wave to recover m(t).
Non-linear practical type (slicer) of PPM receiver:
Function of slicer:
1. It preserves the position of edges of
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received pulses.
2. Removes noise, if any.
Procedure:
1. Slicing level is set approximately to
half the peak of amplitude of PPM wave.

2. Slicer produces rectangular pulses of

sharp leading and trailing edges with an
appropriate delay.

3.Output of slicer is differentiated and

half wave rectified to remove negative
pulses.
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Advantages of PPM

1. Due to constant amplitude of PPM pules, the information is not obtained in the amplitude. Hence
the noise added to the PPM signal does not distort the information. Thus it has good noise
immunity.
2. It is possible to reconstruct PPM signal from the noise contaminated PPM signal.

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3. Due to constant amplitude of pulses, the transmitted power always remains constant. It does not
change as it is used to in PWM.

Disadvantages of PPM

1. As the position of PPM pulses is varied with reference to a reference pulse, a transmitter has to

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send synchronizing pulses to operate the timing circuits in the receiver. Without them the
demodulation won’t be possible to achieve.
2. Large bandwidth is required to ensure transmission of undistorted pulses.

5.6 The Quantization Process

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A continuous signal has infinite number of amplitude levels. It is not necessary in fact to transmit the
exact amplitudes of the samples. Any human sense can detect only finite intensity differences.
The original continuous signal may be approximated by a signal constructed of discrete amplitudes
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selected on a minimum error basis from an available set.

Amplitude quantization is defined as the processes of transforming the sample amplitude m(nTs) of
message signal m(t) at time t = nTs into a discrete amplitude v(nTs) taken from a finite set of possible
amplitudes. We assume that the quantization process is memoryless and instantaneous.
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Fig.10 Description of memory less quantizer

Since we are dealing with a memoryless quantizer, we may use the symbol m in place of
m(nTs).
◊ The signal amplitude m is specified by the index k if it lies inside the interval where , where
L is the total number of amplitude levels used in the quantizer.

................
(5.15)
The mk are called decision levels or decision thresholds. The vk are called representation
levels or reconstruction levels.
The spacing between two adjacent representation levels is called a quantum or step-size.

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The mapping v = g(m) ................ (5.16)

is the quantizer characteristic, which is a staircase function by definition. Quantizers can be
uniform or nonuniform. In a uniform quantizer, the representation levels are uniformly spaced,
otherwise, the quantizer is non-uniform. The quantizer characteristic can be of midtread or
midrise type.

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Fig. 11 Types of Quantization
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5.6.1 Quantization Noise
The use of quantization introduces an error defined as the difference between the input
signal m and the output signal v. This error is called quantization noise. For simplicity, let the
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quantizer input m be the sample value of a zero-mean random variable M. A quantizer Q( )

maps the input random variable M of continuous amplitude into a discrete random variable
V.
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Fig.12 Illustration of Quantization Procss

Let the quantization error be denoted by the random variable Q of sample value q. Thus we
write
q= m–υ
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OR Q = M –V
................ (5.17)
With the input M having zero mean, and the quantizer assumed to be symmetric it follows
that the quantizer output V and therefore the quantization error Q will also have zero mean.
In order to find the output signal-to-noise ratio, we need to find the mean-square value of the
quantization error Q.
Consider an input m of continuous amplitude in the range (-mmax,mmax). Assuming a uniform
quantizer of the midrise type, we find the step-size of the quantizer is given by

................ (5.18)

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where L is the total number of representation levels. For a uniform quantizer, the
quantization error Q will have its sample values bounded by -Δ/2 ≤ q ≤ Δ/2 . If Δ is
sufficiently small or L is sufficiently large, it is reasonable to assume that the quantization
error Q is a uniform distributed random variable. The probability density function of the
quantization error . The probability density function of the quantization error Q is

RA ................ (5.19)
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From these Eqns, ................ (5.20)
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Typically, the L-ary number k, denoting the kth representation level of the quantizer is
transmitted to the receiver in binary form. Let R denote the number of bits per sample used
in the construction of the binary code

Substituting Eq. (7.40) in (7.36) we get the step size

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................ (5.21)
The output signal-to-noise ratio of a uniform quantizer is given by

................ (5.22)
Eq. (7.44) shows that the output SNR of the quantizer increases exponentially with
increasing of bits per sample.

5.7 Pulse-Code Modulation

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In pulse-code modulation (PCM) a message signal is represented by a sequence of coded

pulses, which is accomplished by representing the signal in discrete form in both time and
amplitude. The transmitter section of a PCM circuit consists of Sampling, Quantizing and Encoding,
which are performed in the analog-to-digital converter section. The LPF prior to sampling prevents
aliasing of the message signal. The basic operations in the receiver section are regeneration of
impaired signals, decoding, and reconstruction of the quantized pulse train. Fig.13 is the block
diagram of PCM which represents the basic elements of both the transmitter and the receiver sections.

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Fig. 13 Basic elements of PCM System

Low Pass Filter: This filter eliminates the high frequency components present in the input analog signal
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which is greater than the highest frequency of the message signal, to avoid aliasing of the message
signal.
Sampler: This is the technique which helps to collect the sample data at instantaneous values of
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message signal, so as to reconstruct the original signal. The sampling rate must be greater than twice the
highest frequency component Wof the message signal, in accordance with the sampling theorem.
Quantizer: Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer, reduces the redundant bits and compresses the value.
Uniform quantizing is not the optimum solution because at low signal levels the quantizing noise is high
and the SNR is very low. At high signal levels the quantizing noise is the same even though we would
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tolerate a high noise level. Uniform quantisation was therefore replaced by a non-uniform quantisation
process called companding.
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.
Two major types of compression laws:
µ Law A Law

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m and v are the normalized input and output voltages, respectively. The case of uniform quantization
corresponds to μ = 0 and A = 1.
Encoder: The digitization of analog signal is done by the encoder. It designates each quantized level by
a binary code. The sampling done here is the sample-and-hold process. These three sections (LPF,

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Sampler, and Quantizer) will act as an analog to digital converter. To exploit the advantages of
sampling and quantizing for the purpose of making the transmitted signal more robust to
noise,
interference, and other channel degradations, we require the use of an encoding process to
translate the discrete set of sample values to a more appropriate form of signal. Encoding
minimizes the bandwidth used. Any plan for representing each of this discrete events is called a
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code.
Differential Encoding: This method is used to encode information in terms of signal
transitions. A transition is used to designate symbol 0 in the incoming binary data stream,
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while no transition is used to designate symbol 1.

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Regenerative Repeater: This section (as shown in the fig.14) increases the signal strength. The output
of the channel also has one regenerative repeater circuit, to compensate the signal loss and reconstruct
the signal, and also to increase its strength.
Regenerative repeaters, which perform three basic functions: equalization, timing, and
decision making.

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Fig.14 Block diagram of Regenerative Repeater

Two measure factors: bit error rate (BER) and jitter.

Decoder: The decoder circuit decodes the pulse coded waveform to reproduce the original signal. This

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circuit acts as the demodulator. The decoding process involves generating a pulse the amplitude
of which is the linear sum of all the pulses in the code word, with each pulse being weighted
by its place value (20,21,…,2R-1) ,where R is the number of bits per sample.

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A
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Reconstruction Filter: After the digital-to-analog conversion is done by the regenerative circuit and the
decoder, a low-pass filter is employed, called as the reconstruction filter to get back the original signal.
Hence, the Pulse Code Modulator circuit digitizes the given analog signal, codes it and samples it, and
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then transmits it in an analog form. This whole process is repeated in a reverse pattern to obtain the
original signal.
Filtering: The final operation in the receiver is to recover the message signal wave by passing
the decoder output through a low-pass reconstruction filter whose cut-off frequency is equal to
the
message bandwidth W.

5.8 Application to Vocoders

Vocoders (stands for voice encoder) are analysis and synthesis system, used to reproduce human speech
in limited bandwidth. The vocoder was initially conceived and developed as a communication tool in the
1930s as a means of coding speech for delivery over telephone wires. The vocoder takes the two signals,
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and using their spectral information, creates a third signal. The goal of the vocoder is to imprint the
amplitude and frequency characteristics of the speech signal onto the timbre of the synthesis signal,
whilst maintaining the pitch of the speech signal. Vocoders represent voice signal in the form of bits
using codebook, hence speech data rates reduced.
The basic component extracted during the vocoder analysis is called the formant. The formant describes
the fundamental frequency of a sound and its associated noise components.
Current voice technologies:
1) Global Systems for Mobile communications (GSM): is a digital mobile telephony system. GSM
digitizes and compresses data, then sends it down a channel with two other streams of user data,
each in its own time slot. It operates at either the 900 MHz or 1800 MHz frequency band.
2) Code Division Multiple Access (CDMA) -- supports cellular and data, used in second-generation

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(2G) and third-generation (3G) wireless communications. It allows numerous signals to occupy a single
transmission channel, optimizing the use of available bandwidth. The technology is used in ultra-high-
frequency (UHF) cellular telephone systems in the 800-MHz and 1.9-GHz bands.
3) LTE -- Long Term Evolution is a 4G wireless broadband technology developed to achieve high-speed
data. The first devices that supported LTE technology appeared in 2010.

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4) VoLTE -- Voice over Long Term Evolution, utilizing IMS technology. It is one method for creating,
provisioning and managing high-speed voice, video and messaging services on a 4G wireless network
for mobile and portable devices delivered over IP via an LTE access network.

Idea behind Vocoders

The voice consists of sound made by human being using vocal folds for talking, singing, laughing,
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crying etc. The human voice is specifically a part of human sound production in which vocal folds
(vocal cords) are primary sound source. The mechanism for generating the human voice can be
subdivided into three parts: lungs, vocal folds (within larynx) and articulators (the part of vocal tract
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above the larynx consist of tongue, palate, cheek, lips etc.,).

Basic working of Vocoders:

A vocoder requires two inputs a carrier wave (sound), and a modulator (human voice) input to function
properly. The modulator takes the voice, finds its fundamental frequencies (important bits), and converts
them into levels of amplitude on a series of band pass filters (this is why some vocoders have different
numbers of bands). In general, the more bands available the more understandable the speech will be.
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These band pass filter signals are then passed onto the carrier wave where the final sound is created.

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(a)
Vocoding systems
1. Channel vocoder: The channel vocoder employs a bank of bandpass filters, each having a
bandwidth between 100 Hz and 300Hz. Typically, 16-20 linear phase FIR filter are used. The output

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of each filter is rectified and low pass filtered. The bandwidth of the low pass filter is selected to
match the time variations in the characteristics of the vocal tract as shown in the fig. 15.
At the receiver the signal samples are passed through D/A converters. The outputs of the D/As are
multiplied by the voiced or unvoiced signal sources. The resulting signals are passed through
bandpass filters. The outputs of the bandpass filters are summed to form the synthesized speech
signal.
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(b)

Fig. 15 Block diagram of Channel vocoder (a) Analysis (b) Synthesis

2. Linear predictive coder (LPC): The objective of LP analysis is to estimate parameters of an all-
pole model for the vocal tract.
3. Formant vocoder: The formant vocoder can be viewed as a type of channel vocoder that estimates
the first three or four formants in a segment of speech. It is this information plus the pitch period
that is encoded and transmitted to the receiver.

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Module – 5 Digital Representation of Analog Signals 15EC45

4. Voice excited vocoder. The phase vocoder is similar to the channel vocoder. However, instead of
estimating the pitch, the phase vocoder estimates the phase derivative at the output of each filter. By
coding and transmitting the phase derivative, this vocoder destroys the phase information.

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