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PULSE

MODULATION
PULSE MODULATION

Pulse modulation consists essentially


of sampling analog information signals
and then converting those samples into
discrete pulses and transporting the
pulses from a source to a destination
over a physical transmission medium.
PULSE MODULATION

The four predominant methods of


pulse modulation include:

Pulse Width Modulation (PWM)


Pulse Position Modulation (PPM)
Pulse Amplitude Modulation (PAM)
Pulse Code Modulation (PCM)
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 proportional to
the amplitude of the analog signal at the
time the signal is sampled.
PULSE POSITION MODULATION

With PPM, the position of a constant-


width pulse within a prescribed time slot
is varied according to the amplitude of
the sample of the analog signal. The
higher the amplitude of the sample, the
farther to the right the pulse is positioned
within the prescribed time slot.
PULSE POSITION MODULATION

The highest amplitude sample produces


a pulse to the far right, and the lowest
amplitude sample produces a pulse to the
far left.
PULSE AMPLITUDE MODULATION

The amplitude of a constant width,


constant-position pulse is varied
according to the amplitude of the sample
of the analog signal.The amplitude of a
pulse coincides with the amplitude of the
analog signal. PAM waveforms resemble
the original analog signal more than the
waveforms for PWM or PPM.
PULSE CODE MODULATION

With PCM, the analog signal is sampled


and then converted to a serial n-bit
binary code for transmission. Each code
has the same number of bits and requires
the same length of time for transmission.
PULSE MODULATION
PULSE CODE
MODULATION
PULSE CODE MODULATION

PCM is the only digitally encoded


modulation technique shown in Figure 1
that is commonly used for digital
transmission. The term pulse code
modulation is somewhat of amisnomer,
as it is not really a type of modulation
but rather a form of digitally coding
analog signals.
PULSE CODE MODULATION

With PCM, the pulses are of fixed length


and fixed amplitude. PCM is a binary
system where a pulse or lack of a pulse
within a prescribed time slot represents
either a logic 1 or a logic 0 condition.
PWM, PPM, and PAM are digital but
seldom binary, as a pulse does not
represent a single binary digit (bit).
PULSE CODE MODULATION
PULSE CODE MODULATION
PCM TRANSMITTER
Bandpass filter - limits the frequency of
the analog input signal to
the standard voice-band frequency range
of 300 Hz to 3000 Hz.
Sample-And-Hold circuit - periodically
samples the analog input signal and
converts those samples to a multilevel
PAM signal.
PULSE CODE MODULATION
PCM TRANSMITTER
Analog-to-Digital Converter (ADC) -
converts the PAM samples to parallel
PCM codes, which are converted to
serial binary data in the parallel-to-
serial converter and then outputted onto
the transmission line as serial digital
pulses.
PULSE CODE MODULATION
PCM TRANSMITTER
Transmission Line Repeaters -
are placed at prescribed distances to
regenerate the digital pulses.
PULSE CODE MODULATION
PCM RECEIVER

Serial-to-Parallel Converter - converts


serial pulses received from the
transmission line to parallel PCM codes.

Digital-to-Analog Converter (DAC) -


converts the parallel PCM codes to
multilevel PAM signals.
PULSE CODE MODULATION
PCM RECEIVER

Serial-to-Parallel Converter - converts


serial pulses received from the
transmission line to parallel PCM codes.

Digital-to-Analog Converter (DAC) -


converts the parallel PCM codes to
multilevel PAM signals.
PULSE CODE MODULATION
PCM RECEIVER

Hold Circuit - is basically a lowpass


filter that converts the PAM signals back
to its original analog form.
PULSE CODE MODULATION SAMPLING

The function of a sampling circuit in a


PCM transmitter is to periodically
sample the continually changing analog
input voltage and convert those samples
to a series of constant amplitude pulses
that can more easily be converted to
binary PCM code.
PULSE CODE MODULATION SAMPLING

The function of a sampling circuit in a


PCM transmitter is to periodically
sample the continually changing analog
input voltage and convert those samples
to a series of constant amplitude pulses
that can more easily be converted to
binary PCM code.
PULSE CODE MODULATION SAMPLING

There are two basic techniques used to


perform the sampling function:

• Natural Sampling
• Flat-Top Sampling
PULSE CODE MODULATION SAMPLING

Natural sampling is when tops of the


sample pulses retain their natural shape
during the sample interval, making it
difficult for an ADC to convert the
sample to a PCM code. With natural
sampling, the frequency spectrum of the
sampled output is different from that of
an ideal sample.
PULSE CODE MODULATION SAMPLING

NATURAL SAMPLING
PULSE CODE MODULATION SAMPLING

Flat-top sampling - accomplished in a


sample-and-hold circuit. The purpose of
a sampleand-hold circuit is to
periodically sample the continually
changing analog input voltage and
convert those samples to a series of
constant-amplitude PAM voltage levels.
PULSE CODE MODULATION SAMPLING

With flat-top sampling, the input voltage


is sampled with a narrow pulse and then
held relatively constant until the next
sample is taken.
PULSE CODE MODULATION SAMPLING

FLAT-TOP SAMPLING
SAMPLING RATE

The Nyquist sampling theorem establishes the


minimum sampling rate (fs) that can be used for
a given PCM system. For a sample to be
reproduced accurately in a PCM receiver, each
cycle of the analog input signal (fa) must be
sampled at least twice. Consequently, the
minimum sampling rate is equal to twice the
highest audio input frequency. If fs is less than
two times fa, an impairment called alias or
foldover distortion occurs.
SAMPLING RATE

Mathematically, the minimum Nyquist sampling


rate is

fs ≥ 2fa

where fs = minimum Nyquist sample rate (hertz)


fa = maximum analog input frequency
(hertz)
SAMPLING RATE
ALIASING
SAMPLING RATE
ALIASING

An alias is a signal that is mistakenly


sampled when the sampling frequency is
less than twice the input frequency. An
antialiasing filter is used to ensure that the
correct signal is used.
SAMPLING RATE
ALIASING

To eliminate this problem, a low-pass filter


called an antialiasing filter is usually
placed between the modulating signal
source and the A/D converter input to
ensure that no signal with a frequency
greater than one-half the sampling
frequency is passed.
SAMPLING RATE
ALIASING

Most antialiasing filters use multiple-stage


LC filters, an RC active filter, or high-order
switched capacitor filters to give the steep
roll-off required to eliminate any aliasing.
The filter cutoff is usually set just slightly
above the highest- frequency content of the
input signal.
SAMPLING RATE
ALIASING
A digital communication system uses
sampling at 10kilosamplespersecond
(kSa/s). The receiver filters out all
frequencies above 5 kHz. What frequencies
appear at the receiver for each of the
following signal frequencies at the input to
the transmitter? (a) 1 kHz
(b) 5 kHz
(c) 6 kHz
SAMPLING RATE
ALIASING

It is necessary to transmit the human voice


using a frequency range from 300 Hz to 3.5
kHz using a digital system.
(a) What is the minimum required sampling
rate, according to theory?
(b) Why would a practical system need a
higher rate than the one you calculated in
part (a)
QUANTIZATION

Quantization is the process of converting an


infinite number of possibilities to a finite
number of conditions. Analog signals
contain an infinite number of amplitude
possibilities.

Quantization is the process of rounding off


the amplitudes of flat-top samples to a
manageable number of levels.
QUANTIZATION
QUANTIZATION

Assigning PCM codes to absolute magnitudes is called


quantizing. The magnitude of a quantum is also called
the resolution.

Resolution - is equal to the voltage of the minimum


step size, which is equal to the voltage of the least
significant bit (Vlsb) of the PCM code. The resolution
is the minimum voltage other than 0 V that can be
decoded by the digital-to-analog converter in the
receiver.
QUANTIZATION

Each three-bit code has a range of input voltages that


will be converted to that code. For example, any
voltage between +0.5 and +1.5 will be converted to
the code 101 (+1 V). Each code has a quantization
range equal to + or - one-half the magnitude
of a quantum except the codes for +0 and -0. The 0-V
codes each have an input range equal to only one-half
a quantum (0.5 V).
QUANTIZATION
QUANTIZATION
The figure shows an analog input signal,
the sampling pulse, the corresponding
quantized signal (PAM), and the PCM code
for each sample. The likelihood of a sample
voltage being equal to one of the eight
quantization levels is remote. Therefore, as
shown in the figure, each sample voltage is
rounded off (quantized) to the closest
available level and then converted to its
corresponding PCM code.
QUANTIZATION
Quantization Error (Qe) - any round-off
errors in the transmitted signal that are
reproduced when the code is converted
back to analog in the receiver. Quantization
error is also called quantization noise (Qn).
The maximum magnitude for the
quantization error is equal to onehalf a
quantum.
QUANTIZATION
The first sample shown in Figure at time
t1,when the input voltage is exactly +2 V.
The PCM code that corresponds to +2 V is
110, and there is no quantization error.
Sample 2 occurs at time t2, when the input
voltage is +1 V. The corresponding PCM
code is 001, and again there is no
quantization error.
QUANTIZATION
To determine the PCM code for a particular
sample voltage, simply divide the voltage
by the resolution, convert the quotient to an
n-bit binary code,and then add the sign bit.
For sample 3 in Figure,the voltage at t3 is
approximately +2.6 V.
QUANTIZATION
Example
For the PCM coding scheme shown in
Figure, determine the quantized voltage,
quantization error (Qe), and PCM code for
the analog sample voltage of +1.07 V.
DYNAMIC RANGE
Dynamic range(DR) is the ratio of the
largest possible magnitude to the smallest
possible magnitude (other than 0 V) that
can be decoded by the digital-to-analog
converter in the receiver. Mathematically,
dynamic range is
DR = Vmax/Vmin
DR = dynamic range (unitless ratio)
Vmin = the quantum value (resolution)
Vmax = the maximum voltage magnitude that
can be discerned by the DACs in the receiver
DYNAMIC RANGE
Can be written as:
DR = Vmax/Resolution
Dynamic range is generally expressed as a dB value; therefore,
DR = 20log(Vmax/Vmin)
The number of bits used for a PCM code depends on the dynamic range.
The relationship between dynamic range and the number of bits in a PCM
code is
2n - 1 ≥ DR
and for a minimum number of bits
2n -1 = DR
where
n = number of bits in a PCM code, excluding the sign bit
DR = absolute value of dynamic range
DYNAMIC RANGE
Example
For a PCM system with the following parameters,
determine
(a) minimum sample rate,
(b) minimum number of bits used in the PCM code,
(c) resolution, and
(d) quantization error.

Maximum analog input frequency = 4 kHz


Maximum decoded voltage at the receiver = 2.55 V
Minimum dynamic range = 46 dB
DYNAMIC RANGE
Example
a.) Minimum sample rate = 8kHz
b.) Minimum number of bits used
in the PCM code= 7.63
c.) Resolution = 0.01V
d.) Quantization Error = 0.005V
DYNAMIC RANGE
Example
a.) Minimum sample rate = 8kHz
b.) Minimum number of bits used
in the PCM code= 7.63
c.) Resolution = 0.01V
d.) Quantization Error = 0.005V
CODING EFFICIENCY

Coding efficiency is a numerical indication of how


efficiently a PCM code is utilized. Coding efficiency
is the ratio of the minimum number of bits required to
achieve a certain dynamic range to the actual number
of PCM bits used. Mathematically, coding efficiency
is
minimum number of bits (including sign bit)
coding efficiency = x 100
actual number of bits (including sign bit)
SIGNAL-TO-QUANTIZATION NOISE RATIO

is a measure of the quality of the quantization, or


digital conversion of an analog signal. Defined as
normalized signal power divided by normalized
quantization noise power.

The maximum quantization noise is half the


resolution (quantum value). Therefore, the worst
possible signal voltage-to-quantization noise voltage
ratio (SQR) occurs when the input signal is at its
minimum amplitude (101 or 001). Mathematically, the
worst-case voltage SQR is
SIGNAL-TO-QUANTIZATION NOISE RATIO

Resolution Vlsb
SQNR = SQNR = =2
Qe Vlsb/2

The worst-case (minimum) SQR occurs for the lowest


magnitude quantization voltage ( 
1V). Therefore, the
minimum SQR is
SIGNAL-TO-QUANTIZATION NOISE RATIO

For Linear PCM Codes


LINEAR VERSUS NONLINEAR PCM CODES
Linear Codes
The magnitude change between any two successive
steps is uniform. With linear codes, resolution for
higher amplitude analog signals is the same for lower
amplitude signals.

Non Linear Codes


The step size increases with the amplitude of the input
signal
LINEAR VERSUS NONLINEAR PCM CODES
COMPANDING
Companding is a process of signal compression
and expansion that is used to overcome problems
of distortion and noise in the transmission of
audio signals.

Companding is the process of compression and


then expansion. With companded system, the
higher amplitude analog signals are compressed
(amplified less than lower amplitude signals)
prior to transmission and then expanded
(amplified more than the lower amplitude signals)
in the receiver.
COMPANDING
The data rate is important in telecommunication
because it is directly proportional to the cost of
transmitting the signal. Saving bits is the same as
saving money. Companding is a common
technique for reducing the data rate of audio
signals by making the quantization levels
unequal.
COMPANDING

ANALOG COMPANDING
COMPANDING

In the United States and Japan, μ-law


companding is used. The compression
characteristics for μ-law is

Vout = output voltage


Vm = maximum possible input voltage
Vin = instantaneous value of input voltage
The value of µ is usually 255.
COMPANDING
TWO BASIC TYPES OF COMPANDING