one of the key benefits of digital transmission which uses only logical "ones" and "zeroes",
i.e., tension or no tension, to code the original signal. These benefits soon encouraged
network infrastructure providers and operators to switch over to Pulse Code Modulation
systems, or PCMs.
Analogue speech signals are sampled at a frequency of 8 kHz, converted into a sequence of
"ones" and "zeroes" and transmitted to the distant party. It is not necessary to process the
entire human voice spectrum between 120 Hz and 10 kHz as this would require a lot of
bandwidth. A limited spectrum between 340 Hz and 3.4 kHz is enough to recognize the
speaker and to present his voice comfortably and understandably.
Speech samples are coded with 8 bits, resulting in a data rate of 64 kbps. The Pulse Code
Modulation that is used in ISDN does not apply a linear sort of quantisation. Small signals,
i.e. quiet speech are converted with more accuracy than large signals representing loud
speech.
The G.711 speech codec family defines two major codec types: a μ-Law Codec used in the
Americas and an A-Law type used in Europe. Both codecs apply a logarithmic sampling
method to convert an analogue signal into digital words of 8 bits, but use different scales for
amplitude quantisation. In both cases 28 = 256 different signal values are retrieved per 125
μs time frame. This is sufficient for good speech quality.
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| Digital Speech Transmission |
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