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G.711 is an ITU
standard for the conversion of an analogue voice signal into a
stream of digital messages.[37]
It is a PCM (Pulse Code Modulation) scheme operating at an 8 kHz
sample rate with 8 bits per sample. Then 64kbps data bandwidth
is required (8k * 8 bits = 64kbps). G.711 can encode frequencies
between 0 and 4 kHz, according to the Nyquist theorem, a signal
must be sampled at twice of its highest frequency component.[36]
Normally G.711 is used to encode the 4kHz analogue signal that
defines "toll-quality" speech. G.711 also supports the worst-case
bandwidth for a single voice channel.[37]
PCM
coding:
First, the input
analogue signal is sampled like a series of pulse modulation signals
(Figure 2-21). The heights of these
pulses are from zero to full scale which are divided into discrete
steps called quantisation levels or discrete levels. In this way,
each step can be represented by a series of binary codes, as shown
in Table 2-5. This coded arrangement
of binary pulses is the PCM signal.[39]

Figure 2-21 Quantising
and digitising a signal[38]
Table 2-5 Quantisation
levels with belonging code words
|
Level
|
Code word
|
|
0
|
000
|
|
1
|
001
|
|
2
|
010
|
|
3
|
011
|
|
4
|
100
|
|
5
|
101
|
|
6
|
110
|
|
7
|
111
|
So this PCM encoded
signal in binary form is like this:
010 101 111 111
110 100 010 001 010 100 110 111
For each sample,
its value is just an approximation of the original signal to the
nearest discrete level. Then the reconstructed sampling signal
is distorted. The difference between the original waveform and
the quantised digital signal is called quantising noise. The accuracy
of an approximation is primarily a matter of economics.[38]
So the number of
samples and the number of quantisation levels are the two main
factors which influence the quality of the digitalized signal.
G.711 uses 8 kHz sample rate with 8 bits per sample to create
"toll-quality" digitalized speech.
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