Pulse Modulation & Pulse Code Modulation
Pulse Modulation & Pulse Code Modulation
By
KASULA RAGHU
Assistant Professor
Dept. of E.C.E.,MGIT
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UNIT- IV
Pulse Modulation
&
Pulse Code Modulation
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Contents
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ANALOG PULSE MODULATION SCHEMES
Pulse Modulation:
Statement: A continuous time signal can be represented in its samples and can
(or)
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Figure: Illustration of sampling process. (a) Analog waveform (b) Instantaneously
sampled representation of the analog Signal
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𝑔δ 𝑡 = 𝑔 𝑡 σ∞
𝑛=−∞ δ(𝑡 − 𝑛𝑇𝑠 )
∞
1 𝑛
𝐺δ 𝑓 = 𝐺(𝑓 − )
𝑇𝑠 𝑇𝑠
𝑛=−∞
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𝐺δ 𝑓
Case-1: fs = 2W
………… …………
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𝐺δ 𝑓
Case-2: fs > 2W
………… …………
Let W= 5KHz Then Sampling Frequency will be equal to Greater than fs= 10KHz
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𝐺δ 𝑓
Case-3: fs < 2W
………… …………
Let W= 5KHz Then Sampling Frequency will be equal to Greater than fs < 2W
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Corrective Measures for Aliasing
1. Prior to sampling, a low-pass anti-aliasing filter is used to attenuate those
high frequency components of the signal that are not essential to the
information being conveyed by the signal.
2. The filtered signal is sampled at a rate slightly higher than the Nyquist rate.
Also, it has the beneficial effect of easing the design of the reconstruction
filter used to recover the original signal from its sampled version.
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PULSE AMPLITUDE MODULATION(PAM)
In PAM , the pulses can be flat top type or natural type or ideal
type. Out of these, flat top PAM is widely used because of easy
noise removal.
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Natural Sampling
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Natural Sampling
• The sample and hold circuit consists of two FETs and a capacitor.
• The sampling switch is closed for a short duration by a short pulse applied to
the gate G1 of transistor.
• During this charged period, the Capacitor is quickly to a voltage equal to
instantaneous sample value of incoming signal x(t).
• When the sampling switch is opened for next half of the duration capacitor is
discharged to zero volts.
• Hence the output of circuit consists of a sequence of natural samples.
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PAM GENERATION
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s(t)= c(t)g(t)
𝐴𝑇 ∞ 𝑛𝑇 𝑗2𝜋𝑛𝑡
c(t)= σ𝑛=−∞ 𝑠𝑖𝑛𝑐 exp
𝑇𝑠 𝑇𝑠 𝑇𝑠
𝐴𝑇 ∞ 𝑛𝑇 𝑗2𝜋𝑛𝑡
s 𝑡 = σ𝑛=−∞ 𝑠𝑖𝑛𝑐 exp 𝑔(𝑡)
𝑇𝑠 𝑇𝑠 𝑇𝑠
𝐴𝑇 ∞ 𝑛𝑇 𝑛
S 𝑓 = σ 𝑠𝑖𝑛𝑐 𝐺(𝑓 − )
𝑇𝑠 𝑛=−∞ 𝑇𝑠 𝑇𝑠
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Spectrum of the Natural Sampled Signal
Note1: we Assumed that g(t) contains no frequencies outside –W to W, and sampling rate
1
is greater than the Nyquist rate 2W, so that there is no Aliasing effect. We see that finite
𝑇
𝑠
𝐴𝑇
duration of the sampled pulses is to multiply with the nth Lobe of the spectrum S(f) by
𝑇𝑠
𝑛𝑇
𝑠𝑖𝑛𝑐 . The Original signal g(t) can be recovered from s(t) by passing through ideal
𝑇𝑠
LPF.
Note2: If AT=1 , so that each Rectangular pulse has Unit Area i.e. Pulse duration T
approaches to Zero, then s 𝑓 approaches 𝐺δ 𝑓 . (Ideal Sampling)
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Flat-Top Sampling
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s 𝑡 = σ∞
𝑛=−∞ 𝑔 𝑛𝑇𝑠 ℎ 𝑡 − 𝑛𝑇𝑠
1 0<𝑡<𝑇
1
h(t)= 𝑡 = 0, 𝑡 = 𝑇
2
0 𝑂𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
𝑔δ 𝑡 = σ∞
𝑛=−∞ 𝑔(𝑛𝑇𝑠 )δ(𝑡 − 𝑛𝑇𝑠 )
∞
= −∞ σ∞
n=−∞ g(nTs )δ(t − nTs ) h(t − τ)dτ
∞
= σ∞
n=−∞ g(nTs ) nTs )h(t − τ)dτ
−∞ δ(t −
Using Shifting Property of Delta Function, we Obtain
g δ t * h(t) = σ∞
n=−∞ g(nTs ) h(t − nTs )
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s(t)= g δ t * h(t)
Taking F.T on Both Sides we Get,
S(f) = Gδ f . H(f)
1 ∞ n
S(f) = σn=−∞ G(f − ). H(f)
Ts Ts
1
Finally, g(t) is a band limited signal with Sampling rate Greater than the Nyquist rate. Then
Ts
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Demodulation of PAM
• For PAM signals, demodulation is done using a holding circuit.
• The received PAM signal is first passed through a holding circuit
and then through a lowpass filer.
• Switch S is closed during the arrival of the pulse and is opened at
the end of the pulse.
• Capacitor C is charged to pulse amplitude value and holds this
value during the interval between two pulses.
• Holding circuit output is then passed through a low pass filter to
extract the original signal.
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Advantages, Disadvantages of PAM
Advantages:
• It is the simple process for modulation and demodulation
• Transmitter and receiver circuits are simple and easy to construct.
Disadvantages:
• Bandwidth requirement is high
• Interference of noise is maximum
• Power requirement is high
Applications:
• Used in microcontrollers for generating control signals
• Used as electronic driver for LED lighting
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Pulse Width Modulation(PWM)
In PWM, the width of pulses of carrier pulse train is varied in
proportion with amplitude of modulating signal.
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GENERATION of PWM & PPM
• Disadvantages of PWM
1. Due to the variable pulse width, the pulses have variable power contents. Hence,
the transmission must be powerful enough to handle the maximum width, pulse,
though the average power transmitted can be as low as 50% of this maximum
power.
2. In order to avoid any waveform distortion, the bandwidth required for the PWM
communication is large as compared to bandwidth of PAM.
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PULSE POSITION MODULATION(PPM)
Modulation technique in which position of pulses of carrier pulse train
is varied in accordance with amplitude of modulating signal.
Generation:
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PPM GENERATION
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DETECTION OF PPM
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DETECTION OF PPM
• The circuit consists of S-R flipflop which is set or gives high
output when reference pulses arrive.
• Reference pulses are generated by a reference pulse generator.
• Flip-flop circuit is reset and gives low output at the leading edge
of PPM signal.
• The process repeats and we get PWM pulses at the output of
flip-flop.
• PWM pulses are then demodulated in a PWM demodulator to
get original modulating signal.
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Comparison between PAM, PWM and PPM
S.No 8PAM PWM PPM
2 Bandwidth depends on the Bandwidth depends on the rise Bandwidth depends on the
width of the pulse time of the pulse rise time of the pulse
3 Instantaneous transmitter Instantaneous transmitter Instantaneous transmitter
power varies with the power varies with the amplitude power remains constant with
amplitude of the pulses and the width of the pulses the width of the pulses
4 System complexity is high System complexity is low System complexity is low
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Multiplexing
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Advantages of multiplexing
Applications of Multiplexing
1. Communication system
2. Computer memory
3. Telephone systems
4. TV broadcasting
5. Telemetry
6. Satellites
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Frequency Division Multiplexing
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• The frequency division multiplexing divides the bandwidth of a channel into several logical
sub-channels (individual signal frequencies (or) set of frequency bands).
• Each logical sub-channel is allotted for a different signal frequency. The individual signals
are filtered and then modulated (frequency is shifted), in order to fit exactly into logical
sub-channels.
• In this technique, each logical sub-channel (individual signal frequency) is allotted to each
user. In other words, each user owns a sub-channel.
• Each logical sub-channel is separated by an unused bandwidth called Guard Band to
prevent overlapping of signals. In other words, there exists a frequency gap between two
adjacent signals to prevent signal overlapping. A guard band is a narrow frequency range
that separates two signal frequencies.
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Advantages of Frequency Division Multiplexing (FDM)
1 FDM stands for Frequency Division Multiplexing. TDM stands for Time Division Multiplexing.
2 FDM is an Analog technique TDM is a Digital technique.
3 The communication channel is divided by The communication channel is divided by time.
frequency
4 All signals of different frequencies are transmitted All signals operate with the same frequency are
simultaneously. transmitted at different times.
5 Synchronization is not required Synchronization is required.
6 The bandwidth of the communication channel The bandwidth capacity of the communication channel
should be greater than the combined bandwidth of should be greater than the multiple input signals.
individual signals.
7 FDM requires complex circuitry at the transmitter TDM does not require complex circuitry.
and receiver.
8 In FDM, the problem of crosstalk is severe. In TDM, the problem of crosstalk is not severe.
9 The channel bandwidth is effectively used. The channel bandwidth is wasted.
10 FDM requires Guard bands for its operation. TDM requires sync pulse for its operation.
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11 FDM is used in TV and RADIO broadcasting TDM is used in Pulse code modulation
Digital Communications
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Introduction
• Digital communication is a mode of communication where the information or the thought
is encoded digitally as discreet signals and electronically transferred to the recipients.
• n digital communication information flows in a digital form and the source is generally
the keyboard of the computer. A single individual is capable of digital communication and
thus it also saves wastage of manpower and is one of the cheapest modes of
communication.
• Digital communication is also a really quick way to communicate. The information can
reach the recipient within a fraction of a second. An individual no longer has to wait to
personally meet the other individual and share his information.
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Basic Digital Communication System
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Advantages
• The Digital Communication's it provides us added security to our information signal.
• The digital Communication system has more immunity to noise and external interference.
• Digital information can be saved and retrieved when necessary while it is not possible in analog.
• In Digital Communication System, the error correction and detection techniques can be
implemented easily.
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Disadvantages
• Disadvantages of digital communication:
1) Generally, more bandwidth is required than that for analog systems.
2) Synchronization is required.
3) High power consumption (Due to various stages of conversion).
4) Complex circuit, more sophisticated device making is also drawbacks of digital system.
5) Introduce sampling error
6) As square wave is more affected by noise, That’s why while communicating through
channel we send sin waves but while operating on device we use squire pulses.
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Shannon–Hartley theorem
• The Shannon–Hartley theorem states the channel capacity C, meaning the
theoretical tightest upper bound on the information rate of data that can be
communicated at an arbitrarily low error rate using an average received signal
power S through an analog communication channel subject to additive white
Gaussian noise of power N
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• Where C is the channel capacity in bits per second, a theoretical upper bound
on the net bit rate (information rate, sometimes denoted I) excluding error-
correction codes;
• B is the bandwidth of the channel in hertz (passband bandwidth in case of a
bandpass signal);
• S is the average received signal power over the bandwidth (in case of a carrier-
modulated passband transmission, often denoted C), measured in watts (or
volts squared);
• N is the average power of the noise and interference over the bandwidth,
measured in watts (or volts squared); and
• S/N is the signal-to-noise ratio (SNR) or the carrier-to-noise ratio (CNR) of the
communication signal to the noise and interference at the receiver (expressed
as a linear power ratio, not as logarithmic decibels).
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Analog to Digital Conversion
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Pulse Code Modulation(PCM)
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Sampler
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Block diagram of regenerative repeater
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Regenerative Repeater
A regenerative repeater (see Figure 3.18) consists of (1) an equalizer, (2) a timing circuit, and
(3) a decision-making device. The equalizer is used to undo the effect of the transmission
channel to get back the pulses in their original shape before transmission. The timing circuit
is used to recover the clock of the transmitted symbols (pulses), which is then used in the
decision-making process. The function of the decision-making device is to detect the
different pulses based on some threshold information.
The purpose of a regenerative repeater is to clean the PCM signal during its transmission
through a channel.
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Differential Pulse Code Modulation
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DPCM
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e 𝑛𝑇𝑠 = m 𝑛𝑇𝑠 − 𝑚 ෝ 𝑛𝑇𝑠
It is the difference between unquantized input sample m 𝑛𝑇𝑠 and a prediction of it
𝑚ෝ 𝑛𝑇𝑠 .Prediction value is obtained by using Prediction filter. The difference signal e 𝑛𝑇𝑠 is
called prediction error.
𝑚𝑞 𝑛𝑇𝑠 = 𝑚
ෝ 𝑛𝑇𝑠 +𝑒𝑞 𝑛𝑇𝑠
𝑝
𝑥ො 𝑛 = 𝑚
ෝ 𝑛𝑇𝑠 =σ𝑘=𝑖 𝑤𝑘 𝑚𝑞 𝑛𝑇𝑠 − 𝑘𝑇𝑠
𝑝
e 𝑛𝑇𝑠 = m 𝑛𝑇𝑠 − σ𝑘=𝑖 𝑤𝑘 𝑚𝑞 𝑛𝑇𝑠 − 𝑘𝑇𝑠
The Weights 𝑤𝑘 should be adjusted Properly to have Good Prediction.
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ADVANTAGES OF DPCM
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Delta Modulation / One Bit Modulation
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DM system. (a) Transmitter. (b) Receiver.
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Delta Modulation (DM)
In DM, the message signal is over-sampled to purposely increase correlation
between adjacent samples.
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. Illustration of delta modulation
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Illustration of the two different forms of quantization error in
delta modulation.
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Delta Modulation (Cont’d)
The transmitter of a DM system (Figure 3.23a) is given by a comparator, a one-bit quantizer,
an accumulator, and an encoder.
The receiver of a DM system (Figure 3.23b) is given by a decoder, an accumulator, and a low-
pass filter.
DM is subject to two types of quantization error: Slope overload distortion and granular noise
(see Figure 3.24).
Slope overload distortion is due to the fact that the staircase approximation mq(t) can't follow
closely the actual curve of the message signal m(t). In order for mq(t) to follow closely m(t), it
is required that
dm(t )
max
Ts dt
be satisfied. Otherwise, step-size is too small for the staircase approximation mq(t) to follow
m(t).
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Delta Modulation (Cont’d)
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SQNR of PCM
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Derivation of Maximum Signal to Quantization Noise Ratio for Linear Quantization:
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SNR for DM System
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Adaptive Delta Modulation
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Block Diagram of Adaptive Delta Modulation
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Adaptive delta Modulation
Adaptive delta modulation (ADM) is a modification of DM, in which the step
size is adapted to the slope (variation) of the message signal.
If successive errors are of opposite polarity, then the delta modulator is operating
in the granular mode; in such a case it is advantageous to use reduced step size.
If successive errors are of the same polarity, then the delta modulator is operating
in its slope-overload mode; in this case, the step size should be increased.
The algorithm used for adaptive DM with step size increase/decrease of 50% is
(k)=|(k-1)|(mq(k)+0.5mq(k-1))/mq(k), if (k-1)min
(k)=min, if (k-1)min
• where if (k) is the step size at iteration k and mq(k) is the one-bit quantizer output
that is equal to 1.
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Adaptive delta modulation system: (a) Transmitter. (b) Receiver
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Figure 3.32
Waveforms resulting from the computer experiment on delta
modulation: (a) Linear delta modulation. (b) Adaptive delta
modulation.
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Block diagram illustrating the linear adaptive prediction process.
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Differential PCM (DPCM)
Voice and video signals represented in PCM exhibit high correlation, which means that
PCM signals contain redundant information. The result is an inefficient coding.
By removing the PCM information redundancy a more efficient coded signal may be
obtained. This is done using DPCM.
In DPCM a linear prediction is performed on samples of a message signal m(kTs)=m(k),
then the prediction error e(k ) m(k ) mˆ (k ) is computed and fed to a quantizer to obtain
the quantized value eq(k)=e(k)+q(k), as shown by Figure 3.28a. q(k) is the quantization
error.
The input of the linear predictor of Figure 3.28a is mq (k ) mˆ (k ) eq (k ) m(k ) q (k ) ,
which represents a quantized version of the input sample m(k).
If the prediction is well performed, then the variance of e(k) will be much smaller than
the variance of m(k), which results into a smaller number of levels to quantize e(k).
The receiver as given by Figure 3.28b, consists of a decoder which produces eq(k), that is
added to the output of a prediction filter identical to the one used in the transmitter. The
result is the quantized message signal mq(k).
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Differential PCM (Continued)
DPCM includes DM as a special case, where the prediction filter is a simple delay
element. Simply put, DM is a one-bit version of DPCM.
The problem of slope-overload distortion may also arise in DPCM, whenever the
slope of the message signal changes too rapidly for the prediction filter to track it.
The noise performance of DPCM is measured, as in other digital modulation
systems, by the output signal-to-quantization noise, given by
2
2
2
( SNR )o M2 M2 E G ( SNR )
Q E 2 p Q
Q
where 2M , Q2 , and 2E are the variances of m(k), q(k), and e(k), respectively.
The factor Gp is the processing gain produced by the DPCM quantization scheme.
When Gp>1, which is the case most of the time, it represents the gain in SNR
obtained by using DPCM compared to PCM.
The receiver as given by Figure 3.28b, consists of a decoder which produces eq(k),
that is added to the output of a prediction filter identical to the one used in the
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transmitter. The result is the quantized message signal mq(k).
Figure 3.28
DPCM system. (a)
Transmitter. (b)
Receiver.
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Adaptive DPCM
In PCM, the standard bit rate is 64 kbits/s. The aim of all the variants of
PCM is to reduce the number of bits used in the encoding process by
removing redundancies.
Adaptive DPCM (ADPCM) is a scheme that permits the coding of speech
(voice) signals at 32 kbits/s through the combined use of adaptive
quantization and adaptive prediction.
Adaptive quantization refers to a quantizer that operates with a time-
varying step-size (k ) ˆ M (k ) and adaptive prediction filter refers to a filter
with time-varying coefficients. is a constant and ˆ M (k ) is an estimate of
the standard deviation of m(k).
In ADPCM adaptive quantization can be performed using adaptive
quantization with forward estimation (AQF) or adaptive quantization with
backward
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Adaptive DPCM (Continued)
In ADPCM adaptive prediction can be performed using adaptive prediction with
forward estimation (APF) or adaptive prediction with backward estimation
(APB).
AQF and APF use unquantized samples of the input message signal to estimate
M and the predictor coefficients w, respectively.
AQB and APB use quantized samples of the input message signal to estimate M
and the predictor coefficients w, respectively.
Both AQF and APF suffer from the same disadvantages, which are the buffering,
the side (extra) information to be transmitted, and the delay. But by using AQB
and APB these disadvantages are eliminated.
Figure 3.29 shows the AQB scheme and Figure 3.30 shows the APB scheme.
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Adaptive quantization with backward estimation (AQB).
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Adaptive prediction with backward estimation (APB).
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Line Encoding Techniques
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Digital Data, Digital Signals
[the technique used in a number of LANs]
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ON-OFF Coding
1 Positive Voltage
0 NO Voltage
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Bipolar Signaling /Non Return to Zero
Equal Positive & Negative Amplitude Voltage are represented
1 Positive Voltage
0 Negative Voltage
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Return to Zero (RZ)
1 Half Pulse width Rectangle is used for symbol ‘1’
0 No Voltage
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Split Phase or Manchester encoding
1 For Symbol ‘1’ with Positive pulse followed by negative pulse
with both pulses being of equal amplitude and half-symbol wide.
0 For Symbol ‘0’ the Polarities of these pulses are reversed.
(Negative pulse followed by Positive pulse)
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Differential Encoding
1 No Transition for Symbol ‘1’
0 Transition for Symbol ‘0’ based on the Reference Bit at Starting.
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Bit rate and Baud Rate
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Prob1 :
An analog signal carries 4 bits/symbol elements. If 1000 signal elements are sent per
second, Find bit rate and total number of elements.
Sol: n=4 bits/symbol elements
r= 1000 signal elements /second Bit Rate R= nr
=4x1000 =4kbps
The total number of elements L= 2n=24 =16
Prob2 :
An analog signal has a bit rate of 8000bps and a baud rate of 1000 baud. How many data elements
are carried by each signal element? How many signal elements do we need?
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Non-Uniform Quantization
• In non-uniform quantization, the step size is not fixed. It varies according to certain law or
as per input signal amplitude. The following fig shows the characteristics of Nonuniform
quantizer.
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Companding PCM System
Non-uniform quantizers are difficult to make and expensive.
An alternative is to first pass the speech signal through nonlinearity before quantizing
with a uniform quantizer.
The nonlinearity causes the signal amplitude to be compressed.
At the receiver, the signal is expanded by an inverse to the nonlinearity.
The process of compressing and expanding is called Companding.
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Companding
This is a non-linear technique / Companding 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.
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µ-law Companding Technique.
A- L aw Companding:
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µ-law Companding Technique
For given value of µ ,reciprocal slope of compression curve
defines Quantum steps
dx/dy = log(1+µ)/µ
= 1+µ|x|
A-law Companding Technique:
For given value of A ,reciprocal slope of compression curve
defines Quantum steps
dx/dy = (1+logA)/A 0≤|x|≤1/A
= (1+logA)|x| 1/A≤|x|≤1
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Thank You
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