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Part 05
Pulse Code Modulation
Pulse Code Modulation
(PCM)
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Advantages of Digital Communication
Digital systems are less sensitive to noise and signal distortion. For long transmission line, the signal
may be regenerated effectively error‐free at different points along the path.
With digital systems, it is easier to integrate different services, e.g., video and the accompanying
soundtrack into the same transmission scheme
soundtrack, into the same transmission scheme.
The transmission scheme can be relatively independent of the source
Circuitry for digital signals is easier to repeat and digital circuits are less sensitive to physical effects
such as vibration and temperature
p
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 design easier.
Various media sharing strategies (known as multiplexing) are more easily implemented with digital
t
transmission strategies
i i t t i
Source coding techniques can be used for removing redundancy from digital transmission
Error‐control coding can be used for adding redundancy, which can be used to detect and correct
errors at the receiver side
errors at the receiver side
Digital communication systems can be made highly secure by exploiting powerful encryption algorithms
Digital communication systems are inherently more efficient than analog communication systems in
the tradeoff between transmission bandwidth and signal‐to‐noise ratio
Various channel compensation techniques, such as, channel estimation and equalization, are easier to
implement
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Sampling (1)
Sampling is an operation that is basic to digital signal processing and digital communications
Through the use of sampling process, an analog signal is converted into a corresponding
q p y p y
sequence of samples that are usually spaced uniformly in time
Message
Sampling Signal
T f f s
s f nf
s
n
Sampled Signal
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Sampling (2)
Two types of practical sampling:
Natural Sampling
Flat‐top sampling
Fl t t li
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Sampling (3)
Frequency Domain:
or,
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Sampling (4)
fs > 2W:
fs = 2W:
fs < 2W:
Aliasing
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Sampling Theorem
Sampling theorem is a fundamental bridge between continuous signals (analog
domain) and discrete signals (digital domain)
It only applies to a class of mathematical functions whose Fourier transforms are
zero outside of a finite region of frequencies
fs = Sampling frequency
fs = 2W: Nyquist frequency / Nyquist rate / Minimum sampling frequency
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Antialiasing Filter
All practical signals are time-limited, i.e., non band-limited => Aliasing inevitable
To limit aliasing
aliasing, use antialiasing filter (LPF) before sampling
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Reconstruction Filter
fs = 2W:
Ideal LPF
‐ 1/2W 1/2W
characteristic:
(interpolation filter /
interpolation function)
Ts = 1/2W
(interpolation formula)
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Quantization (1)
It is the process of transforming the sample amplitude m(nTS) of a baseband signal at time t = nTS
into a discrete amplitude v(nTS) taken from a finite set of possible levels
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Quantization (2)
Quantizer characteristic:
kth interval:
Here, k = 1, 2, 3, … , L
L = Number of representation levels
(Number of intervals)
mk: Decision levels / Decision thresholds
vk: Representation levels / Reconstruction levels / Quantization Levels
Δ=|vk +1 – vk | = |mk +1 – mk|: Step-size / quantum
Quantizer output equals to vk if the input signal sample m belongs to the interval Ik (rounding)
v vk if m I k (R di )
(Rounding)
v vk if vk m vk 1 (Truncation)
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Quantization(3): Two types
Mid‐tread quantization Mid‐rise quantization
Representation/
Reconstruction/
Quantization
Q
levels
Mid‐tread quantizer:
q Mid‐rise quantizer:
Reconstruction value is exactly zero Decision threshold value is exactly zero
Signal Range (Dynamic range) and Quantizer Range: Could be same or different
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Quantization(4): Example
For the following sequence {1.2, -0.2, -0.5, 0.4, 0.89,1.3} quantize it using a
uniform quantizer of rounding type and write the quantized sequence.
Quantizer range is (-1.5,1.5) with 4 levels.
Solution:
Thus, 1.2 fall between 0.75 and 1.5, and hence is quantized to 1.125.
Quantized sequence:
{1.125, -0.375, -0.375, 0.375, 1.125, 1.125}
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Quantization(5): Two types
Uniform quantization Non‐uniform quantization 15
Quantization Error for Uniform Quantization (1)
Quantization error (noise)
q = m – v => Q = M – V
fQ(q)
1/Δ
Quantization noise power
/2
2
q f Q q dq
2
Q
2
– Δ/2 0 Δ/2
/ 2
12
q
P 12 P
12P
Signal-to-nose-ratio (SNR):SNR P = Average power of m(t)
Q2 2
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Quantization Error for Uniform Quantization (2)
Suppose m(t) of continuous amplitude in the range [‐mmax, mmax]:
2mmax 2mmax R = Number of bits for presenting each level (bits/sample)
L 2R
3P 2 R 3P
SNR 2 2 SNRdB 6.02 R 10 log 2
mmax mmax
Each additional bit increases the SNR by 6.02 dB and
a corresponding increase in required channel BW
Special case:
m(t) is a sinusoidal signal with amplitude equal to mmax
3
SNR 2 2 R SNRdB 6 R 1.8
2
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Non-Uniform Quantization
SNR of weak signals is much lower than that of strong signal
SNR of weak signals is much lower than that of strong signal
Instantaneous SNR is also lower for the smaller amplitudes compared to that of the
larger amplitudes
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Non-Uniform Quantization
‐ Step size increases as the separation from the origin of the input–output amplitude
characteristic is increased
‐ First Compression and then uniform quantization
and then uniform quantization
‐ Achieve more even SNR over the
dynamic range using fewer bits (e.g.,
8 bits instead of 13/14 bits)
Receiver side: Expansion required
Compression + Expansion = Companding
di
Rx
Tx
Original Uniform Original
Compression Reconstruction Expansion
Signal Quantization Signal
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Non-Uniform Quantization
The compression here occurs in the amplitude values
Compression in amplitudes means that the amplitudes of the compressed signal
are more closelyy spaced
p in comparison
p to the original
g signal
g
To do so, the compressor boosts the small amplitudes by a large amount. However,
the large amplitude values receive very small gain and the maximum value remains
the same
Compressor Input Compressor Output
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Non-Uniform Quantization
μ‐Law: Used in North America, Japan (μ = 255 is mostly used)
More uniform SNR is achieved over a larger dynamic range
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Non-Uniform Quantization
A‐Law: Used in Europe and many other countries
A = 87.6 is mostly used and comparable to μ = 255
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Encoding
Each quantized samples is encoded into a code word
Each element in a code word is called code element
Binary code:
Each code element is either of two distinct
values, customarily denoted as 0 and 1
Binary symbol withstands a relatively high
level of noise and also easy to regenerate
Each binary code word consists of R bits and
hence, this code can represent 2R distinct
numbers (i.e.,
(i e at best R bit quantizer can be
used)
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Pulse Code Modulation (PCM)
In 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
Transmitter
Three basic operations in a PCM Transmitter:
‐ Sampling
‐ Quantization
‐ Encoding
Transmission
Path
Receiver
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Differential PCM (DPCM)
When a signal is sampled at a rate slightly higher than the Nyquist rate, there exists a
high degree correlation between adjacent samples, i.e., in an average sense, the signal
does not change rapidly from one sample to the next
When these highly correlated samples are encoded as in a standard PCM system
system, the
resulting encoded signal contains redundant information implying that symbols that are not
absolutely essential to the transmission of information are generated
DPCM removes this redundancy y before encodingg byy taking
g the difference between the
actual sample m(nTS) and its predicted value m̂nTS
The quantized version of the prediction error e(nTS) are encoded instead of encoding
the samples of the original signal
This will result in much smaller quantization intervals leading to less quantization noise
and much higher SNR
Transmitter
Prediction error
enTS mnTS mˆ nTS
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Predictor for DPCM:
Liner predictor of order p:
Transversal filter (tapped-delay-line filter) used as a linear predictor
p
m̂nTS wk mq n k TS
k 1
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Differential PCM (DPCM)
T
Transmitter
itt
eq nTS
m' nTS
Receiver
m̂nTS m ' nnTS mˆ nnTS eq nnTS
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Delta Modulation (DM)…(2)
Transmitter
Receiver
‐ Digital equivalent of integration
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Predictor for DM
Note:
(1) DPCM uses a higher order filter.
(2) DM uses a 1st order (p=1) predictor with w1 = 1. Thus, the predicted output is the previous sample.
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Delta Modulation (DM)…(3)
Two types of quantization error:
(1) Slope overload distortion/noise (2) Granular noise
mq(t)
eq(nTS)
Comments:
(1) For avoiding slope overload distortion: larger Δ is desired
An optimal step size (Δ) has to be chosen for minimum overall noise
Example: f
mt Am cos mt | m t |max m Am fs Am max s
Ts m
fs
A
r 2 800
r
m max Voice
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Line Coding (1)
PCM, DPCM and DM are different strategies for source
encoding, which converts an analog signal into digital form
Or NRZ-L
Various line coding
(binary) methods:
Or RZ-AMI
(f) Split-phase
or Manchester
(0 means transition)
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Line Coding (2)
Book:
Polar NRZ / Digital Communications:
Fundamentals and Applications
- Bernard Sklar
Applications:
Polar NRZ / NRZ-L: Digital
g logic
g circuits
NRZ-M/NRZ-S: Magnetic tap recording
RZ line codes: Base band transmission and magnetic recording (e.g., Bipolar RZ / RZ-AMI
is used for telephone system)
Manchester Coding: Magnetic recording, optical communications and satellite telemetry
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Line Coding (3)
Desired properties (i.e., design criteria) for line coding:
Transmission bandwidth: should be as small as possible
Noise immunity: should be immune to noise
Power efficiency: for a given bandwidth and given error probability, transmission power
requirement should be as small as possible
Error detection and correction capability: should be possible to detect and correct errors
Favorable power spectral density (PSD): should have zero PSD at zero (i.e., DC)
frequency, otherwise the ac coupling and the transformers used in communication systems
would block the DC component
Adequate timing information / self-clocking: should carry the timing or clock information
which can be used for self-synchronization
Transparency: should be possible to transmit a digital signal correctly regardless of the
patterns of 1’s and 0’s (by preventing long string of 0s and 1s)
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