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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A brief aside about ADCs

ADCs are used to convert an analogue input voltage into a number that can be interpreted as a physical parameter by a computer.
0111 0110 0100 0101 0011 0010 0001 0000 1111 1110 1100 1010 1101 1011 1001 0000 0110 0111 0011 1100 1001 1011 68

Resolution= 1 part in 2n

Numbers passed from ADC to computer to represent analogue voltage

Sampling
The input signal is sampled prior to digitisation and an approximation to the input is reconstructed by the digital-toanalogue converter:
input

Sampling

Digitisation

code, modulate
Transmission Wire/optical fibre Aerial/free-space

Filtering

Digital-to-analogue

conversion

Demodulate, Decode

output

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Sampling an analogue signal

Prior to digitisation, signals must be sampled
ADC converts the height of each pulse into binary representation Sampling involves the multiplication of the signal by a train of sampling pulses
With a frequency fs=2B=1/T

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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? 73 What is the minimum value of fs for which there is no overlap of the Harmonics with the baseband?

 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 75 rate is 2B samples/s

 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 77

 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 lower82 level than with analogue amplification

 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 Slot 1 Channel 1 Sample 1 Slot 2 Channel 2 Sample 1 Frame n+1 Slot 1 Channel 1 Sample 2 Slot 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? 125 s

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
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87

 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. 88

89

Spectrum of a train of pulses:

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