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Pulse Code Modulation

The advantages of digital communication systems
(cf. analogue communication)
Easier to store as a pattern of 1's and 0's
Increased Immunity



non-linearities
Easier to process in computers and digital signal
processors
Can be coded for security and error correction purposes
Several digital signals can easily be interleaved
(multiplexed) and transmitted on one channel
Noisy digital signals can be regenerated more effectively
than analogue signals can be amplified.
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0000
1111
1110
1101
1100
1011
1010
1001
0001
0010
0011
0100
0101
0110
0111
A brief aside about ADCs
0000 0110 0111 0011 1100 1001 1011
Numbers passed from ADC to computer to represent analogue voltage
ADCs are used to convert an analogue input voltage into a number that can
be interpreted as a physical parameter by a computer.
Resolution=
1 part in 2
n
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Sampling
The input signal is sampled prior to digitisation and an
approximation to the input is reconstructed by the digital-to-
analogue converter:
Sampling Digitisation code, modulate
Transmission
Wire/optical fibre
Aerial/free-space

input
Filtering
Digital-to-analogue
conversion
Demodulate, Decode
output
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Sampling an analogue signal
Prior to digitisation, signals must be sampled
With a frequency f
s
=2B=1/T
ADC converts the height of each pulse into binary representation
Sampling involves the multiplication of the signal by a train of sampling
pulses
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Sampling as multiplication by a sampling
waveform:
Sampling pulse is
short enough so that
can normally
considered have zero
duration
DAC, however
produces pulses
length T
Multiplication = Amplitude modulation
Amplitude modulation produces sidebands
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Sidebands produced by multiplication with a
carrier
That is, amplitude modulation
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Sidebands at each harmonic of the sampling pulse
Digital-to-analogue conversion involves recovery of the baseband
How?
What is the minimum value of f
s
for which there is no overlap of the Harmonics with
the baseband?
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If the sidebands do not overlap the signal
can be recovered
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Practical sampling
the "Sample-and-hold" system:
This is Nyquists theorem
For a signal of bandwidth B Hz, the minimum sampling
rate is 2B samples/s
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Effect of sampling rate
sampling at more than the Nyquist Rate
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Sampling at the Nyquist Rate
cannot build an ideal filter -
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Undersampling
produces aliasing distortion!
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Aliasing-time domain
Oversampled signal
Reconstructed signal
Undersampled signal
Reconstructed signal
Sampling:aliasing & Nyquist:time domain
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The Anti-alias
(Pre-sampling)
filter
ensures that
sampling
obeys the
Nyquist
theorem
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Examples
For the compact disc (Audio CD) the
maximum signal frequency is 20 kHz and the
sampling rate is 44.1 kHz.
The Nyquist Sampling Rate is 40 kHz
Hence the guard band is 4.1 kHz wide.
In the telephone system (see Section 5.8),
the speech signal has a bandwidth up to 3.4
kHz and a sampling rate of 8 kHz,
The Nyquist Sampling Rate is 6.8 kHz
Hence the guard band is 1.2 kHz wide.
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Regeneration v amplification:
Gain of amplifiers equals loss in transmission lines
SNR analog: S/kN
SNR digital: S/N
In practice finite S/N means there will be a low level of bit errors
Some accumulation of bit-error noise with repeaters, but much lower level
than with analogue amplification
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A Pulse-Code Modulation communication
system
"PCM"
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A digital communication system - "PCM"

Anti-alias Filter*
Digitiser/Sample-and-Hold circuit*
Analogue-to-Digital Converter*
Coding-
Source coding for data compression,
Line coding for signalling efficiency
Error coding to reduce the effect of errors
Modulator
Physical Channel (with repeaters if necessary)*
Copper cables
Fibre Optic cables
Radio
Sonar
Recording medium
Demodulator
Decoder (Source-, Line- and Error-)
Digital-to-Analogue Converter*
Reconstruction Filter*
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Time-division Multiplexing "TDM"
Allocate interleaved time-slots to each signal
Assemble the binary coded samples into Frames:
2-channel time-division multiplexing scheme:
Frame n Frame n+1
Slot 1 Slot 2 Slot 1 Slot 2
Channel 1
Sample 1
Channel 2
Sample 1
Channel 1
Sample 2
Channel 2
Sample 2
Two channels share a single physical channel
Cost?
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The 32-channel PCM Transmission system

30 speech signals plus two control channels
for signalling and synchronising:
Signal bandwidth 3.4 kHz
Sampling rate 8 kHz
Hence frame length?
Sample size 8 bits/sample
Hence bit rate from each signal 64 kbit/s
32 channels
Hence each time slot 3.906 s
1/(8000*32)
Overall data rate 2.048 Mbit/s
8000*32*8
125 s
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A number of frames can be time-division multiplexed
together in a TDM heirachy.
4 frames of 32 channels
= 128 basic PCM channels,
Has data rate of 4 x 2.048 Mbit/s = 8.192 Mbit/s
8.448Mbit/s including extra signalling bits

4 x 128 = 512 channels
Has data rate = 4 x8.192 Mbit/s (+ signalling bits)
= 34.368 Mbit/s

etc

Up to a multiplex of 32768 channels with an overall data
rate of 2.48832 Gbit/s.
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Spectrum of a train of pulses: