1. CS 4594 Data Communications
PCM

2. Pulse Code Modulation PCM = Pulse Code Modulation
Converts analog signal to digital signal
Analog signal is limited (filtered) to a range of frequencies
Samples the analog signal at regular intervals
Each sample is converted to a number that represents its amplitude
May use non-linear scale or differential coding

3. PCM and the Telephone System
Limits signal to less than 3400 Hz
Samples at 8000 Hz (Nyquist says to sample at least twice the frequency)
Uses 12 bits to digitize samples
Compresses this to 7 or 8 bits per sample (depending on exact method, ulaw or Alaw)
This is 56 to 64 kbits/sec called DS0

4. Steps of Telephone PCM voice resulting voice signal Linearly quantize
to 12 bits
band limit
frequency
to <= 3400 Hz
Sample
8000
per sec
Compress
to 8 bits
per sample
voice
Trans-
mission
line
resulting
voice
signal
Filter (fills in between pulses)
Pulse generation
(amplitude of pulses match digital values)
Decompress
(back to 12-bit)

5. -Law, A-law Companding
Companding methods
Companding = compressing/expanding
Samples lower amplitudes more closely than higher amplitudes
-law used in North America, etc.
A-law used in Europe, etc.

6. Companding
A-law
-law
Output
signal
Input
signal

7. Variations on PCM
Differential PCM - send difference in quantized values as 2 or 3 bit value
Adaptive DPCM - varies quantization levels according to rate of change of input signal
ITU-T has
ADPCM standard for 4 kHZ voice at 40, 32, 24, and 16 kbits/sec.
ADPCM standard for 7 kHZ voice at 64 kbits/sec